Adding upstream version 124.0.1.upstream/124.0.1

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

66 files changed, 15754 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/BUILD.gn b/third_party/libwebrtc/audio/BUILD.gn
new file mode 100644
index 0000000000..ec09e5a350
--- /dev/null
+++ b/third_party/libwebrtc/audio/BUILD.gn
@@ -0,0 +1,250 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio") {
+  sources = [
+    "audio_level.cc",
+    "audio_level.h",
+    "audio_receive_stream.cc",
+    "audio_receive_stream.h",
+    "audio_send_stream.cc",
+    "audio_send_stream.h",
+    "audio_state.cc",
+    "audio_state.h",
+    "audio_transport_impl.cc",
+    "audio_transport_impl.h",
+    "channel_receive.cc",
+    "channel_receive.h",
+    "channel_receive_frame_transformer_delegate.cc",
+    "channel_receive_frame_transformer_delegate.h",
+    "channel_send.cc",
+    "channel_send.h",
+    "channel_send_frame_transformer_delegate.cc",
+    "channel_send_frame_transformer_delegate.h",
+    "conversion.h",
+    "remix_resample.cc",
+    "remix_resample.h",
+  ]
+
+  deps = [
+    "../api:array_view",
+    "../api:call_api",
+    "../api:field_trials_view",
+    "../api:frame_transformer_interface",
+    "../api:function_view",
+    "../api:rtp_headers",
+    "../api:rtp_parameters",
+    "../api:scoped_refptr",
+    "../api:sequence_checker",
+    "../api:transport_api",
+    "../api/audio:aec3_factory",
+    "../api/audio:audio_frame_api",
+    "../api/audio:audio_frame_processor",
+    "../api/audio:audio_mixer_api",
+    "../api/audio_codecs:audio_codecs_api",
+    "../api/crypto:frame_decryptor_interface",
+    "../api/crypto:frame_encryptor_interface",
+    "../api/crypto:options",
+    "../api/neteq:neteq_api",
+    "../api/rtc_event_log",
+    "../api/task_queue",
+    "../api/task_queue:pending_task_safety_flag",
+    "../api/transport/rtp:rtp_source",
+    "../api/units:time_delta",
+    "../call:audio_sender_interface",
+    "../call:bitrate_allocator",
+    "../call:call_interfaces",
+    "../call:rtp_interfaces",
+    "../common_audio",
+    "../common_audio:common_audio_c",
+    "../logging:rtc_event_audio",
+    "../logging:rtc_stream_config",
+    "../media:media_channel",
+    "../media:rtc_media_base",
+    "../modules/async_audio_processing",
+    "../modules/audio_coding",
+    "../modules/audio_coding:audio_coding_module_typedefs",
+    "../modules/audio_coding:audio_encoder_cng",
+    "../modules/audio_coding:audio_network_adaptor_config",
+    "../modules/audio_coding:red",
+    "../modules/audio_device",
+    "../modules/audio_processing",
+    "../modules/audio_processing:api",
+    "../modules/audio_processing:audio_frame_proxies",
+    "../modules/audio_processing:rms_level",
+    "../modules/pacing",
+    "../modules/rtp_rtcp",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
+    "../rtc_base:audio_format_to_string",
+    "../rtc_base:buffer",
+    "../rtc_base:checks",
+    "../rtc_base:event_tracer",
+    "../rtc_base:logging",
+    "../rtc_base:macromagic",
+    "../rtc_base:race_checker",
+    "../rtc_base:rate_limiter",
+    "../rtc_base:refcount",
+    "../rtc_base:rtc_event",
+    "../rtc_base:rtc_numerics",
+    "../rtc_base:rtc_task_queue",
+    "../rtc_base:safe_conversions",
+    "../rtc_base:safe_minmax",
+    "../rtc_base:stringutils",
+    "../rtc_base:threading",
+    "../rtc_base:timeutils",
+    "../rtc_base/containers:flat_set",
+    "../rtc_base/experiments:field_trial_parser",
+    "../rtc_base/synchronization:mutex",
+    "../rtc_base/system:no_unique_address",
+    "../rtc_base/task_utils:repeating_task",
+    "../system_wrappers",
+    "../system_wrappers:field_trial",
+    "../system_wrappers:metrics",
+    "utility:audio_frame_operations",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+if (rtc_include_tests) {
+  rtc_library("audio_end_to_end_test") {
+    testonly = true
+
+    sources = [
+      "test/audio_end_to_end_test.cc",
+      "test/audio_end_to_end_test.h",
+    ]
+    deps = [
+      ":audio",
+      "../api:simulated_network_api",
+      "../api/task_queue",
+      "../call:fake_network",
+      "../call:simulated_network",
+      "../modules/audio_device:audio_device_api",
+      "../modules/audio_device:test_audio_device_module",
+      "../system_wrappers",
+      "../test:test_common",
+      "../test:test_support",
+      "../test:video_test_constants",
+    ]
+  }
+
+  rtc_library("audio_tests") {
+    testonly = true
+
+    sources = [
+      "audio_receive_stream_unittest.cc",
+      "audio_send_stream_tests.cc",
+      "audio_send_stream_unittest.cc",
+      "audio_state_unittest.cc",
+      "channel_receive_frame_transformer_delegate_unittest.cc",
+      "channel_send_frame_transformer_delegate_unittest.cc",
+      "channel_send_unittest.cc",
+      "mock_voe_channel_proxy.h",
+      "remix_resample_unittest.cc",
+      "test/audio_stats_test.cc",
+      "test/nack_test.cc",
+      "test/non_sender_rtt_test.cc",
+    ]
+    deps = [
+      ":audio",
+      ":audio_end_to_end_test",
+      ":channel_receive_unittest",
+      "../api:libjingle_peerconnection_api",
+      "../api:mock_audio_mixer",
+      "../api:mock_frame_decryptor",
+      "../api:mock_frame_encryptor",
+      "../api:scoped_refptr",
+      "../api/audio:audio_frame_api",
+      "../api/audio_codecs:audio_codecs_api",
+      "../api/audio_codecs:builtin_audio_encoder_factory",
+      "../api/audio_codecs/opus:audio_decoder_opus",
+      "../api/audio_codecs/opus:audio_encoder_opus",
+      "../api/crypto:frame_decryptor_interface",
+      "../api/rtc_event_log",
+      "../api/task_queue:default_task_queue_factory",
+      "../api/task_queue/test:mock_task_queue_base",
+      "../api/units:time_delta",
+      "../api/units:timestamp",
+      "../call:mock_bitrate_allocator",
+      "../call:mock_call_interfaces",
+      "../call:mock_rtp_interfaces",
+      "../call:rtp_interfaces",
+      "../call:rtp_receiver",
+      "../call:rtp_sender",
+      "../common_audio",
+      "../logging:mocks",
+      "../modules/audio_device:audio_device_api",
+      "../modules/audio_device:audio_device_impl",  # For TestAudioDeviceModule
+      "../modules/audio_device:mock_audio_device",
+      "../modules/audio_mixer:audio_mixer_impl",
+      "../modules/audio_mixer:audio_mixer_test_utils",
+      "../modules/audio_processing:audio_processing_statistics",
+      "../modules/audio_processing:mocks",
+      "../modules/pacing",
+      "../modules/rtp_rtcp:mock_rtp_rtcp",
+      "../modules/rtp_rtcp:rtp_rtcp_format",
+      "../rtc_base:checks",
+      "../rtc_base:gunit_helpers",
+      "../rtc_base:macromagic",
+      "../rtc_base:refcount",
+      "../rtc_base:rtc_base_tests_utils",
+      "../rtc_base:safe_compare",
+      "../rtc_base:task_queue_for_test",
+      "../rtc_base:threading",
+      "../rtc_base:timeutils",
+      "../system_wrappers",
+      "../test:audio_codec_mocks",
+      "../test:field_trial",
+      "../test:mock_frame_transformer",
+      "../test:mock_transformable_frame",
+      "../test:mock_transport",
+      "../test:rtp_test_utils",
+      "../test:run_loop",
+      "../test:scoped_key_value_config",
+      "../test:test_common",
+      "../test:test_support",
+      "../test:video_test_constants",
+      "../test/time_controller:time_controller",
+      "utility:utility_tests",
+      "//testing/gtest",
+    ]
+  }
+
+  rtc_library("channel_receive_unittest") {
+    testonly = true
+    sources = [ "channel_receive_unittest.cc" ]
+    deps = [
+      ":audio",
+      "../api/audio_codecs:builtin_audio_decoder_factory",
+      "../api/crypto:frame_decryptor_interface",
+      "../api/task_queue:default_task_queue_factory",
+      "../logging:mocks",
+      "../modules/audio_device:audio_device_api",
+      "../modules/audio_device:mock_audio_device",
+      "../modules/rtp_rtcp",
+      "../modules/rtp_rtcp:rtp_rtcp_format",
+      "../rtc_base:logging",
+      "../rtc_base:threading",
+      "../test:audio_codec_mocks",
+      "../test:mock_transport",
+      "../test:test_support",
+      "../test/time_controller",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+}
diff --git a/third_party/libwebrtc/audio/DEPS b/third_party/libwebrtc/audio/DEPS
new file mode 100644
index 0000000000..7a0c7e7ce6
--- /dev/null
+++ b/third_party/libwebrtc/audio/DEPS
@@ -0,0 +1,27 @@
+include_rules = [
+  "+call",
+  "+common_audio",
+  "+logging/rtc_event_log",
+  "+media/base",
+  "+modules/async_audio_processing",
+  "+modules/audio_coding",
+  "+modules/audio_device",
+  "+modules/audio_mixer",
+  "+modules/audio_processing",
+  "+modules/audio_processing/include",
+  "+modules/bitrate_controller",
+  "+modules/congestion_controller",
+  "+modules/pacing",
+  "+modules/rtp_rtcp",
+  "+modules/utility",
+  "+system_wrappers",
+]
+
+specific_include_rules = {
+  "audio_send_stream.cc": [
+    "+modules/audio_coding/codecs/cng/audio_encoder_cng.h",
+  ],
+  "audio_transport_impl.h": [
+    "+modules/audio_processing/typing_detection.h",
+  ]
+}
diff --git a/third_party/libwebrtc/audio/OWNERS b/third_party/libwebrtc/audio/OWNERS
new file mode 100644
index 0000000000..e629bc1815
--- /dev/null
+++ b/third_party/libwebrtc/audio/OWNERS
@@ -0,0 +1,5 @@
+alessiob@webrtc.org
+gustaf@webrtc.org
+henrik.lundin@webrtc.org
+jakobi@webrtc.org
+peah@webrtc.org
diff --git a/third_party/libwebrtc/audio/audio_gn/moz.build b/third_party/libwebrtc/audio/audio_gn/moz.build
new file mode 100644
index 0000000000..e81a4f673b
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_gn/moz.build
@@ -0,0 +1,249 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/audio/audio_level.cc",
+    "/third_party/libwebrtc/audio/audio_receive_stream.cc",
+    "/third_party/libwebrtc/audio/audio_send_stream.cc",
+    "/third_party/libwebrtc/audio/audio_state.cc",
+    "/third_party/libwebrtc/audio/audio_transport_impl.cc",
+    "/third_party/libwebrtc/audio/channel_receive.cc",
+    "/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.cc",
+    "/third_party/libwebrtc/audio/channel_send.cc",
+    "/third_party/libwebrtc/audio/remix_resample.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "GLESv2",
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_gn")
diff --git a/third_party/libwebrtc/audio/audio_level.cc b/third_party/libwebrtc/audio/audio_level.cc
new file mode 100644
index 0000000000..7874b73f1c
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_level.cc
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_level.h"
+
+#include "api/audio/audio_frame.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+namespace voe {
+
+AudioLevel::AudioLevel()
+    : abs_max_(0), count_(0), current_level_full_range_(0) {}
+
+AudioLevel::~AudioLevel() {}
+
+void AudioLevel::Reset() {
+  MutexLock lock(&mutex_);
+  abs_max_ = 0;
+  count_ = 0;
+  current_level_full_range_ = 0;
+  total_energy_ = 0.0;
+  total_duration_ = 0.0;
+}
+
+int16_t AudioLevel::LevelFullRange() const {
+  MutexLock lock(&mutex_);
+  return current_level_full_range_;
+}
+
+void AudioLevel::ResetLevelFullRange() {
+  MutexLock lock(&mutex_);
+  abs_max_ = 0;
+  count_ = 0;
+  current_level_full_range_ = 0;
+}
+
+double AudioLevel::TotalEnergy() const {
+  MutexLock lock(&mutex_);
+  return total_energy_;
+}
+
+double AudioLevel::TotalDuration() const {
+  MutexLock lock(&mutex_);
+  return total_duration_;
+}
+
+void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
+  // Check speech level (works for 2 channels as well)
+  int16_t abs_value =
+      audioFrame.muted()
+          ? 0
+          : WebRtcSpl_MaxAbsValueW16(
+                audioFrame.data(),
+                audioFrame.samples_per_channel_ * audioFrame.num_channels_);
+
+  // Protect member access using a lock since this method is called on a
+  // dedicated audio thread in the RecordedDataIsAvailable() callback.
+  MutexLock lock(&mutex_);
+
+  if (abs_value > abs_max_)
+    abs_max_ = abs_value;
+
+  // Update level approximately 9 times per second, assuming audio frame
+  // duration is approximately 10 ms. (The update frequency is every
+  // 11th (= |kUpdateFrequency+1|) call: 1000/(11*10)=9.09..., we should
+  // probably change this behavior, see https://crbug.com/webrtc/10784).
+  if (count_++ == kUpdateFrequency) {
+    current_level_full_range_ = abs_max_;
+
+    count_ = 0;
+
+    // Decay the absolute maximum (divide by 4)
+    abs_max_ >>= 2;
+  }
+
+  // See the description for "totalAudioEnergy" in the WebRTC stats spec
+  // (https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy)
+  // for an explanation of these formulas. In short, we need a value that can
+  // be used to compute RMS audio levels over different time intervals, by
+  // taking the difference between the results from two getStats calls. To do
+  // this, the value needs to be of units "squared sample value * time".
+  double additional_energy =
+      static_cast<double>(current_level_full_range_) / INT16_MAX;
+  additional_energy *= additional_energy;
+  total_energy_ += additional_energy * duration;
+  total_duration_ += duration;
+}
+
+}  // namespace voe
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/audio_level.h b/third_party/libwebrtc/audio/audio_level.h
new file mode 100644
index 0000000000..acd1231fe2
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_level.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_AUDIO_LEVEL_H_
+#define AUDIO_AUDIO_LEVEL_H_
+
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioFrame;
+namespace voe {
+
+// This class is thread-safe. However, TotalEnergy() and TotalDuration() are
+// related, so if you call ComputeLevel() on a different thread than you read
+// these values, you still need to use lock to read them as a pair.
+class AudioLevel {
+ public:
+  AudioLevel();
+  ~AudioLevel();
+  void Reset();
+
+  // Returns the current audio level linearly [0,32767], which gets updated
+  // every "kUpdateFrequency+1" call to ComputeLevel() based on the maximum
+  // audio level of any audio frame, decaying by a factor of 1/4 each time
+  // LevelFullRange() gets updated.
+  // Called on "API thread(s)" from APIs like VoEBase::CreateChannel(),
+  // VoEBase::StopSend().
+  int16_t LevelFullRange() const;
+  void ResetLevelFullRange();
+  // See the description for "totalAudioEnergy" in the WebRTC stats spec
+  // (https://w3c.github.io/webrtc-stats/#dom-rtcaudiohandlerstats-totalaudioenergy)
+  // In our implementation, the total audio energy increases by the
+  // energy-equivalent of LevelFullRange() at the time of ComputeLevel(), rather
+  // than the energy of the samples in that specific audio frame. As a result,
+  // we may report a higher audio energy and audio level than the spec mandates.
+  // TODO(https://crbug.com/webrtc/10784): We should either do what the spec
+  // says or update the spec to match our implementation. If we want to have a
+  // decaying audio level we should probably update both the spec and the
+  // implementation to reduce the complexity of the definition. If we want to
+  // continue to have decaying audio we should have unittests covering the
+  // behavior of the decay.
+  double TotalEnergy() const;
+  double TotalDuration() const;
+
+  // Called on a native capture audio thread (platform dependent) from the
+  // AudioTransport::RecordedDataIsAvailable() callback.
+  // In Chrome, this method is called on the AudioInputDevice thread.
+  void ComputeLevel(const AudioFrame& audioFrame, double duration);
+
+ private:
+  enum { kUpdateFrequency = 10 };
+
+  mutable Mutex mutex_;
+
+  int16_t abs_max_ RTC_GUARDED_BY(mutex_);
+  int16_t count_ RTC_GUARDED_BY(mutex_);
+  int16_t current_level_full_range_ RTC_GUARDED_BY(mutex_);
+
+  double total_energy_ RTC_GUARDED_BY(mutex_) = 0.0;
+  double total_duration_ RTC_GUARDED_BY(mutex_) = 0.0;
+};
+
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_LEVEL_H_
diff --git a/third_party/libwebrtc/audio/audio_receive_stream.cc b/third_party/libwebrtc/audio/audio_receive_stream.cc
new file mode 100644
index 0000000000..c49b83f95f
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_receive_stream.cc
@@ -0,0 +1,475 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_receive_stream.h"
+
+#include <string>
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/call/audio_sink.h"
+#include "api/rtp_parameters.h"
+#include "api/sequence_checker.h"
+#include "audio/audio_send_stream.h"
+#include "audio/audio_state.h"
+#include "audio/channel_receive.h"
+#include "audio/conversion.h"
+#include "call/rtp_config.h"
+#include "call/rtp_stream_receiver_controller_interface.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const {
+  char ss_buf[1024];
+  rtc::SimpleStringBuilder ss(ss_buf);
+  ss << "{remote_ssrc: " << remote_ssrc;
+  ss << ", local_ssrc: " << local_ssrc;
+  ss << ", nack: " << nack.ToString();
+  ss << ", rtcp_event_observer: "
+     << (rtcp_event_observer ? "(rtcp_event_observer)" : "nullptr");
+  ss << '}';
+  return ss.str();
+}
+
+std::string AudioReceiveStreamInterface::Config::ToString() const {
+  char ss_buf[1024];
+  rtc::SimpleStringBuilder ss(ss_buf);
+  ss << "{rtp: " << rtp.ToString();
+  ss << ", rtcp_send_transport: "
+     << (rtcp_send_transport ? "(Transport)" : "null");
+  if (!sync_group.empty()) {
+    ss << ", sync_group: " << sync_group;
+  }
+  ss << '}';
+  return ss.str();
+}
+
+namespace {
+std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive(
+    Clock* clock,
+    webrtc::AudioState* audio_state,
+    NetEqFactory* neteq_factory,
+    const webrtc::AudioReceiveStreamInterface::Config& config,
+    RtcEventLog* event_log) {
+  RTC_DCHECK(audio_state);
+  internal::AudioState* internal_audio_state =
+      static_cast<internal::AudioState*>(audio_state);
+  return voe::CreateChannelReceive(
+      clock, neteq_factory, internal_audio_state->audio_device_module(),
+      config.rtcp_send_transport, event_log, config.rtp.local_ssrc,
+      config.rtp.remote_ssrc, config.jitter_buffer_max_packets,
+      config.jitter_buffer_fast_accelerate, config.jitter_buffer_min_delay_ms,
+      config.enable_non_sender_rtt, config.decoder_factory,
+      config.codec_pair_id, std::move(config.frame_decryptor),
+      config.crypto_options, std::move(config.frame_transformer),
+      config.rtp.rtcp_event_observer);
+}
+}  // namespace
+
+AudioReceiveStreamImpl::AudioReceiveStreamImpl(
+    Clock* clock,
+    PacketRouter* packet_router,
+    NetEqFactory* neteq_factory,
+    const webrtc::AudioReceiveStreamInterface::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    webrtc::RtcEventLog* event_log)
+    : AudioReceiveStreamImpl(clock,
+                             packet_router,
+                             config,
+                             audio_state,
+                             event_log,
+                             CreateChannelReceive(clock,
+                                                  audio_state.get(),
+                                                  neteq_factory,
+                                                  config,
+                                                  event_log)) {}
+
+AudioReceiveStreamImpl::AudioReceiveStreamImpl(
+    Clock* clock,
+    PacketRouter* packet_router,
+    const webrtc::AudioReceiveStreamInterface::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    webrtc::RtcEventLog* event_log,
+    std::unique_ptr<voe::ChannelReceiveInterface> channel_receive)
+    : config_(config),
+      audio_state_(audio_state),
+      source_tracker_(clock),
+      channel_receive_(std::move(channel_receive)) {
+  RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl: " << config.rtp.remote_ssrc;
+  RTC_DCHECK(config.decoder_factory);
+  RTC_DCHECK(config.rtcp_send_transport);
+  RTC_DCHECK(audio_state_);
+  RTC_DCHECK(channel_receive_);
+
+  RTC_DCHECK(packet_router);
+  // Configure bandwidth estimation.
+  channel_receive_->RegisterReceiverCongestionControlObjects(packet_router);
+
+  // When output is muted, ChannelReceive will directly notify the source
+  // tracker of "delivered" frames, so RtpReceiver information will continue to
+  // be updated.
+  channel_receive_->SetSourceTracker(&source_tracker_);
+
+  // Complete configuration.
+  // TODO(solenberg): Config NACK history window (which is a packet count),
+  // using the actual packet size for the configured codec.
+  channel_receive_->SetNACKStatus(config.rtp.nack.rtp_history_ms != 0,
+                                  config.rtp.nack.rtp_history_ms / 20);
+  channel_receive_->SetReceiveCodecs(config.decoder_map);
+  // `frame_transformer` and `frame_decryptor` have been given to
+  // `channel_receive_` already.
+}
+
+AudioReceiveStreamImpl::~AudioReceiveStreamImpl() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_LOG(LS_INFO) << "~AudioReceiveStreamImpl: " << remote_ssrc();
+  Stop();
+  channel_receive_->SetAssociatedSendChannel(nullptr);
+  channel_receive_->ResetReceiverCongestionControlObjects();
+}
+
+void AudioReceiveStreamImpl::RegisterWithTransport(
+    RtpStreamReceiverControllerInterface* receiver_controller) {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  RTC_DCHECK(!rtp_stream_receiver_);
+  rtp_stream_receiver_ = receiver_controller->CreateReceiver(
+      remote_ssrc(), channel_receive_.get());
+}
+
+void AudioReceiveStreamImpl::UnregisterFromTransport() {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  rtp_stream_receiver_.reset();
+}
+
+void AudioReceiveStreamImpl::ReconfigureForTesting(
+    const webrtc::AudioReceiveStreamInterface::Config& config) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+
+  // SSRC can't be changed mid-stream.
+  RTC_DCHECK_EQ(remote_ssrc(), config.rtp.remote_ssrc);
+  RTC_DCHECK_EQ(local_ssrc(), config.rtp.local_ssrc);
+
+  // Configuration parameters which cannot be changed.
+  RTC_DCHECK_EQ(config_.rtcp_send_transport, config.rtcp_send_transport);
+  // Decoder factory cannot be changed because it is configured at
+  // voe::Channel construction time.
+  RTC_DCHECK_EQ(config_.decoder_factory, config.decoder_factory);
+
+  // TODO(solenberg): Config NACK history window (which is a packet count),
+  // using the actual packet size for the configured codec.
+  RTC_DCHECK_EQ(config_.rtp.nack.rtp_history_ms, config.rtp.nack.rtp_history_ms)
+      << "Use SetUseTransportCcAndNackHistory";
+
+  RTC_DCHECK(config_.decoder_map == config.decoder_map) << "Use SetDecoderMap";
+  RTC_DCHECK_EQ(config_.frame_transformer, config.frame_transformer)
+      << "Use SetDepacketizerToDecoderFrameTransformer";
+
+  config_ = config;
+}
+
+void AudioReceiveStreamImpl::Start() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (playing_) {
+    return;
+  }
+  channel_receive_->StartPlayout();
+  playing_ = true;
+  audio_state()->AddReceivingStream(this);
+}
+
+void AudioReceiveStreamImpl::Stop() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!playing_) {
+    return;
+  }
+  channel_receive_->StopPlayout();
+  playing_ = false;
+  audio_state()->RemoveReceivingStream(this);
+}
+
+bool AudioReceiveStreamImpl::IsRunning() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return playing_;
+}
+
+void AudioReceiveStreamImpl::SetDepacketizerToDecoderFrameTransformer(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_receive_->SetDepacketizerToDecoderFrameTransformer(
+      std::move(frame_transformer));
+}
+
+void AudioReceiveStreamImpl::SetDecoderMap(
+    std::map<int, SdpAudioFormat> decoder_map) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  config_.decoder_map = std::move(decoder_map);
+  channel_receive_->SetReceiveCodecs(config_.decoder_map);
+}
+
+void AudioReceiveStreamImpl::SetNackHistory(int history_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_GE(history_ms, 0);
+
+  if (config_.rtp.nack.rtp_history_ms == history_ms)
+    return;
+
+  config_.rtp.nack.rtp_history_ms = history_ms;
+  // TODO(solenberg): Config NACK history window (which is a packet count),
+  // using the actual packet size for the configured codec.
+  channel_receive_->SetNACKStatus(history_ms != 0, history_ms / 20);
+}
+
+void AudioReceiveStreamImpl::SetNonSenderRttMeasurement(bool enabled) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  config_.enable_non_sender_rtt = enabled;
+  channel_receive_->SetNonSenderRttMeasurement(enabled);
+}
+
+void AudioReceiveStreamImpl::SetFrameDecryptor(
+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
+  // TODO(bugs.webrtc.org/11993): This is called via WebRtcAudioReceiveStream,
+  // expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_receive_->SetFrameDecryptor(std::move(frame_decryptor));
+}
+
+webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats(
+    bool get_and_clear_legacy_stats) const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  webrtc::AudioReceiveStreamInterface::Stats stats;
+  stats.remote_ssrc = remote_ssrc();
+
+  webrtc::CallReceiveStatistics call_stats =
+      channel_receive_->GetRTCPStatistics();
+  // TODO(solenberg): Don't return here if we can't get the codec - return the
+  //                  stats we *can* get.
+  auto receive_codec = channel_receive_->GetReceiveCodec();
+  if (!receive_codec) {
+    return stats;
+  }
+
+  stats.payload_bytes_received = call_stats.payload_bytes_received;
+  stats.header_and_padding_bytes_received =
+      call_stats.header_and_padding_bytes_received;
+  stats.packets_received = call_stats.packetsReceived;
+  stats.packets_lost = call_stats.cumulativeLost;
+  stats.nacks_sent = call_stats.nacks_sent;
+  stats.capture_start_ntp_time_ms = call_stats.capture_start_ntp_time_ms_;
+  stats.last_packet_received = call_stats.last_packet_received;
+  stats.codec_name = receive_codec->second.name;
+  stats.codec_payload_type = receive_codec->first;
+  int clockrate_khz = receive_codec->second.clockrate_hz / 1000;
+  if (clockrate_khz > 0) {
+    stats.jitter_ms = call_stats.jitterSamples / clockrate_khz;
+  }
+  stats.delay_estimate_ms = channel_receive_->GetDelayEstimate();
+  stats.audio_level = channel_receive_->GetSpeechOutputLevelFullRange();
+  stats.total_output_energy = channel_receive_->GetTotalOutputEnergy();
+  stats.total_output_duration = channel_receive_->GetTotalOutputDuration();
+  stats.estimated_playout_ntp_timestamp_ms =
+      channel_receive_->GetCurrentEstimatedPlayoutNtpTimestampMs(
+          rtc::TimeMillis());
+
+  // Get jitter buffer and total delay (alg + jitter + playout) stats.
+  auto ns = channel_receive_->GetNetworkStatistics(get_and_clear_legacy_stats);
+  stats.packets_discarded = ns.packetsDiscarded;
+  stats.fec_packets_received = ns.fecPacketsReceived;
+  stats.fec_packets_discarded = ns.fecPacketsDiscarded;
+  stats.jitter_buffer_ms = ns.currentBufferSize;
+  stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
+  stats.total_samples_received = ns.totalSamplesReceived;
+  stats.concealed_samples = ns.concealedSamples;
+  stats.silent_concealed_samples = ns.silentConcealedSamples;
+  stats.concealment_events = ns.concealmentEvents;
+  stats.jitter_buffer_delay_seconds =
+      static_cast<double>(ns.jitterBufferDelayMs) /
+      static_cast<double>(rtc::kNumMillisecsPerSec);
+  stats.jitter_buffer_emitted_count = ns.jitterBufferEmittedCount;
+  stats.jitter_buffer_target_delay_seconds =
+      static_cast<double>(ns.jitterBufferTargetDelayMs) /
+      static_cast<double>(rtc::kNumMillisecsPerSec);
+  stats.jitter_buffer_minimum_delay_seconds =
+      static_cast<double>(ns.jitterBufferMinimumDelayMs) /
+      static_cast<double>(rtc::kNumMillisecsPerSec);
+  stats.inserted_samples_for_deceleration = ns.insertedSamplesForDeceleration;
+  stats.removed_samples_for_acceleration = ns.removedSamplesForAcceleration;
+  stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
+  stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
+  stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);
+  stats.secondary_discarded_rate = Q14ToFloat(ns.currentSecondaryDiscardedRate);
+  stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
+  stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
+  stats.jitter_buffer_flushes = ns.packetBufferFlushes;
+  stats.delayed_packet_outage_samples = ns.delayedPacketOutageSamples;
+  stats.relative_packet_arrival_delay_seconds =
+      static_cast<double>(ns.relativePacketArrivalDelayMs) /
+      static_cast<double>(rtc::kNumMillisecsPerSec);
+  stats.interruption_count = ns.interruptionCount;
+  stats.total_interruption_duration_ms = ns.totalInterruptionDurationMs;
+
+  auto ds = channel_receive_->GetDecodingCallStatistics();
+  stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
+  stats.decoding_calls_to_neteq = ds.calls_to_neteq;
+  stats.decoding_normal = ds.decoded_normal;
+  stats.decoding_plc = ds.decoded_neteq_plc;
+  stats.decoding_codec_plc = ds.decoded_codec_plc;
+  stats.decoding_cng = ds.decoded_cng;
+  stats.decoding_plc_cng = ds.decoded_plc_cng;
+  stats.decoding_muted_output = ds.decoded_muted_output;
+
+  stats.last_sender_report_timestamp_ms =
+      call_stats.last_sender_report_timestamp_ms;
+  stats.last_sender_report_remote_timestamp_ms =
+      call_stats.last_sender_report_remote_timestamp_ms;
+  stats.sender_reports_packets_sent = call_stats.sender_reports_packets_sent;
+  stats.sender_reports_bytes_sent = call_stats.sender_reports_bytes_sent;
+  stats.sender_reports_reports_count = call_stats.sender_reports_reports_count;
+  stats.round_trip_time = call_stats.round_trip_time;
+  stats.round_trip_time_measurements = call_stats.round_trip_time_measurements;
+  stats.total_round_trip_time = call_stats.total_round_trip_time;
+
+  return stats;
+}
+
+void AudioReceiveStreamImpl::SetSink(AudioSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_receive_->SetSink(sink);
+}
+
+void AudioReceiveStreamImpl::SetGain(float gain) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_receive_->SetChannelOutputVolumeScaling(gain);
+}
+
+bool AudioReceiveStreamImpl::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_receive_->SetBaseMinimumPlayoutDelayMs(delay_ms);
+}
+
+int AudioReceiveStreamImpl::GetBaseMinimumPlayoutDelayMs() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_receive_->GetBaseMinimumPlayoutDelayMs();
+}
+
+std::vector<RtpSource> AudioReceiveStreamImpl::GetSources() const {
+  return source_tracker_.GetSources();
+}
+
+AudioMixer::Source::AudioFrameInfo
+AudioReceiveStreamImpl::GetAudioFrameWithInfo(int sample_rate_hz,
+                                              AudioFrame* audio_frame) {
+  AudioMixer::Source::AudioFrameInfo audio_frame_info =
+      channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
+  if (audio_frame_info != AudioMixer::Source::AudioFrameInfo::kError &&
+      !audio_frame->packet_infos_.empty()) {
+    source_tracker_.OnFrameDelivered(audio_frame->packet_infos_);
+  }
+  return audio_frame_info;
+}
+
+int AudioReceiveStreamImpl::Ssrc() const {
+  return remote_ssrc();
+}
+
+int AudioReceiveStreamImpl::PreferredSampleRate() const {
+  return channel_receive_->PreferredSampleRate();
+}
+
+uint32_t AudioReceiveStreamImpl::id() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return remote_ssrc();
+}
+
+absl::optional<Syncable::Info> AudioReceiveStreamImpl::GetInfo() const {
+  // TODO(bugs.webrtc.org/11993): This is called via RtpStreamsSynchronizer,
+  // expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_receive_->GetSyncInfo();
+}
+
+bool AudioReceiveStreamImpl::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+                                                    int64_t* time_ms) const {
+  // Called on video capture thread.
+  return channel_receive_->GetPlayoutRtpTimestamp(rtp_timestamp, time_ms);
+}
+
+void AudioReceiveStreamImpl::SetEstimatedPlayoutNtpTimestampMs(
+    int64_t ntp_timestamp_ms,
+    int64_t time_ms) {
+  // Called on video capture thread.
+  channel_receive_->SetEstimatedPlayoutNtpTimestampMs(ntp_timestamp_ms,
+                                                      time_ms);
+}
+
+bool AudioReceiveStreamImpl::SetMinimumPlayoutDelay(int delay_ms) {
+  // TODO(bugs.webrtc.org/11993): This is called via RtpStreamsSynchronizer,
+  // expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_receive_->SetMinimumPlayoutDelay(delay_ms);
+}
+
+void AudioReceiveStreamImpl::AssociateSendStream(
+    internal::AudioSendStream* send_stream) {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  channel_receive_->SetAssociatedSendChannel(
+      send_stream ? send_stream->GetChannel() : nullptr);
+  associated_send_stream_ = send_stream;
+}
+
+void AudioReceiveStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) {
+  // TODO(solenberg): Tests call this function on a network thread, libjingle
+  // calls on the worker thread. We should move towards always using a network
+  // thread. Then this check can be enabled.
+  // RTC_DCHECK(!thread_checker_.IsCurrent());
+  channel_receive_->ReceivedRTCPPacket(packet, length);
+}
+
+void AudioReceiveStreamImpl::SetSyncGroup(absl::string_view sync_group) {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  config_.sync_group = std::string(sync_group);
+}
+
+void AudioReceiveStreamImpl::SetLocalSsrc(uint32_t local_ssrc) {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  // TODO(tommi): Consider storing local_ssrc in one place.
+  config_.rtp.local_ssrc = local_ssrc;
+  channel_receive_->OnLocalSsrcChange(local_ssrc);
+}
+
+uint32_t AudioReceiveStreamImpl::local_ssrc() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  RTC_DCHECK_EQ(config_.rtp.local_ssrc, channel_receive_->GetLocalSsrc());
+  return config_.rtp.local_ssrc;
+}
+
+const std::string& AudioReceiveStreamImpl::sync_group() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return config_.sync_group;
+}
+
+const AudioSendStream*
+AudioReceiveStreamImpl::GetAssociatedSendStreamForTesting() const {
+  RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
+  return associated_send_stream_;
+}
+
+internal::AudioState* AudioReceiveStreamImpl::audio_state() const {
+  auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
+  RTC_DCHECK(audio_state);
+  return audio_state;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/audio_receive_stream.h b/third_party/libwebrtc/audio/audio_receive_stream.h
new file mode 100644
index 0000000000..db49631638
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_receive_stream.h
@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_AUDIO_RECEIVE_STREAM_H_
+#define AUDIO_AUDIO_RECEIVE_STREAM_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/audio/audio_mixer.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/rtp_headers.h"
+#include "api/sequence_checker.h"
+#include "audio/audio_state.h"
+#include "call/audio_receive_stream.h"
+#include "call/syncable.h"
+#include "modules/rtp_rtcp/source/source_tracker.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+class PacketRouter;
+class RtcEventLog;
+class RtpStreamReceiverControllerInterface;
+class RtpStreamReceiverInterface;
+
+namespace voe {
+class ChannelReceiveInterface;
+}  // namespace voe
+
+namespace internal {
+class AudioSendStream;
+}  // namespace internal
+
+class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStreamInterface,
+                                     public AudioMixer::Source,
+                                     public Syncable {
+ public:
+  AudioReceiveStreamImpl(
+      Clock* clock,
+      PacketRouter* packet_router,
+      NetEqFactory* neteq_factory,
+      const webrtc::AudioReceiveStreamInterface::Config& config,
+      const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+      webrtc::RtcEventLog* event_log);
+  // For unit tests, which need to supply a mock channel receive.
+  AudioReceiveStreamImpl(
+      Clock* clock,
+      PacketRouter* packet_router,
+      const webrtc::AudioReceiveStreamInterface::Config& config,
+      const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+      webrtc::RtcEventLog* event_log,
+      std::unique_ptr<voe::ChannelReceiveInterface> channel_receive);
+
+  AudioReceiveStreamImpl() = delete;
+  AudioReceiveStreamImpl(const AudioReceiveStreamImpl&) = delete;
+  AudioReceiveStreamImpl& operator=(const AudioReceiveStreamImpl&) = delete;
+
+  // Destruction happens on the worker thread. Prior to destruction the caller
+  // must ensure that a registration with the transport has been cleared. See
+  // `RegisterWithTransport` for details.
+  // TODO(tommi): As a further improvement to this, performing the full
+  // destruction on the network thread could be made the default.
+  ~AudioReceiveStreamImpl() override;
+
+  // Called on the network thread to register/unregister with the network
+  // transport.
+  void RegisterWithTransport(
+      RtpStreamReceiverControllerInterface* receiver_controller);
+  // If registration has previously been done (via `RegisterWithTransport`) then
+  // `UnregisterFromTransport` must be called prior to destruction, on the
+  // network thread.
+  void UnregisterFromTransport();
+
+  // webrtc::AudioReceiveStreamInterface implementation.
+  void Start() override;
+  void Stop() override;
+  bool IsRunning() const override;
+  void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+  void SetDecoderMap(std::map<int, SdpAudioFormat> decoder_map) override;
+  void SetNackHistory(int history_ms) override;
+  void SetNonSenderRttMeasurement(bool enabled) override;
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override;
+
+  webrtc::AudioReceiveStreamInterface::Stats GetStats(
+      bool get_and_clear_legacy_stats) const override;
+  void SetSink(AudioSinkInterface* sink) override;
+  void SetGain(float gain) override;
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
+  int GetBaseMinimumPlayoutDelayMs() const override;
+  std::vector<webrtc::RtpSource> GetSources() const override;
+
+  // AudioMixer::Source
+  AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
+                                       AudioFrame* audio_frame) override;
+  int Ssrc() const override;
+  int PreferredSampleRate() const override;
+
+  // Syncable
+  uint32_t id() const override;
+  absl::optional<Syncable::Info> GetInfo() const override;
+  bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+                              int64_t* time_ms) const override;
+  void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
+                                         int64_t time_ms) override;
+  bool SetMinimumPlayoutDelay(int delay_ms) override;
+
+  void AssociateSendStream(internal::AudioSendStream* send_stream);
+  void DeliverRtcp(const uint8_t* packet, size_t length);
+
+  void SetSyncGroup(absl::string_view sync_group);
+
+  void SetLocalSsrc(uint32_t local_ssrc);
+
+  uint32_t local_ssrc() const;
+
+  uint32_t remote_ssrc() const override {
+    // The remote_ssrc member variable of config_ will never change and can be
+    // considered const.
+    return config_.rtp.remote_ssrc;
+  }
+
+  // Returns a reference to the currently set sync group of the stream.
+  // Must be called on the packet delivery thread.
+  const std::string& sync_group() const;
+
+  const AudioSendStream* GetAssociatedSendStreamForTesting() const;
+
+  // TODO(tommi): Remove this method.
+  void ReconfigureForTesting(
+      const webrtc::AudioReceiveStreamInterface::Config& config);
+
+ private:
+  internal::AudioState* audio_state() const;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
+  // TODO(bugs.webrtc.org/11993): This checker conceptually represents
+  // operations that belong to the network thread. The Call class is currently
+  // moving towards handling network packets on the network thread and while
+  // that work is ongoing, this checker may in practice represent the worker
+  // thread, but still serves as a mechanism of grouping together concepts
+  // that belong to the network thread. Once the packets are fully delivered
+  // on the network thread, this comment will be deleted.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_{
+      SequenceChecker::kDetached};
+  webrtc::AudioReceiveStreamInterface::Config config_;
+  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+  SourceTracker source_tracker_;
+  const std::unique_ptr<voe::ChannelReceiveInterface> channel_receive_;
+  AudioSendStream* associated_send_stream_
+      RTC_GUARDED_BY(packet_sequence_checker_) = nullptr;
+
+  bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
+
+  std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_
+      RTC_GUARDED_BY(packet_sequence_checker_);
+};
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_RECEIVE_STREAM_H_
diff --git a/third_party/libwebrtc/audio/audio_receive_stream_unittest.cc b/third_party/libwebrtc/audio/audio_receive_stream_unittest.cc
new file mode 100644
index 0000000000..451d5f9b91
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_receive_stream_unittest.cc
@@ -0,0 +1,431 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_receive_stream.h"
+
+#include <map>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/test/mock_audio_mixer.h"
+#include "api/test/mock_frame_decryptor.h"
+#include "audio/conversion.h"
+#include "audio/mock_voe_channel_proxy.h"
+#include "call/rtp_stream_receiver_controller.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/time_utils.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder_factory.h"
+#include "test/mock_transport.h"
+#include "test/run_loop.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using ::testing::_;
+using ::testing::FloatEq;
+using ::testing::NiceMock;
+using ::testing::Return;
+
+AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
+  AudioDecodingCallStats audio_decode_stats;
+  audio_decode_stats.calls_to_silence_generator = 234;
+  audio_decode_stats.calls_to_neteq = 567;
+  audio_decode_stats.decoded_normal = 890;
+  audio_decode_stats.decoded_neteq_plc = 123;
+  audio_decode_stats.decoded_codec_plc = 124;
+  audio_decode_stats.decoded_cng = 456;
+  audio_decode_stats.decoded_plc_cng = 789;
+  audio_decode_stats.decoded_muted_output = 987;
+  return audio_decode_stats;
+}
+
+const uint32_t kRemoteSsrc = 1234;
+const uint32_t kLocalSsrc = 5678;
+const int kJitterBufferDelay = -7;
+const int kPlayoutBufferDelay = 302;
+const unsigned int kSpeechOutputLevel = 99;
+const double kTotalOutputEnergy = 0.25;
+const double kTotalOutputDuration = 0.5;
+const int64_t kPlayoutNtpTimestampMs = 5678;
+
+const CallReceiveStatistics kCallStats = {678, 234, -12, 567, 78, 890, 123};
+const std::pair<int, SdpAudioFormat> kReceiveCodec = {
+    123,
+    {"codec_name_recv", 96000, 0}};
+const NetworkStatistics kNetworkStats = {
+    /*currentBufferSize=*/123,
+    /*preferredBufferSize=*/456,
+    /*jitterPeaksFound=*/false,
+    /*totalSamplesReceived=*/789012,
+    /*concealedSamples=*/3456,
+    /*silentConcealedSamples=*/123,
+    /*concealmentEvents=*/456,
+    /*jitterBufferDelayMs=*/789,
+    /*jitterBufferEmittedCount=*/543,
+    /*jitterBufferTargetDelayMs=*/123,
+    /*jitterBufferMinimumDelayMs=*/222,
+    /*insertedSamplesForDeceleration=*/432,
+    /*removedSamplesForAcceleration=*/321,
+    /*fecPacketsReceived=*/123,
+    /*fecPacketsDiscarded=*/101,
+    /*packetsDiscarded=*/989,
+    /*currentExpandRate=*/789,
+    /*currentSpeechExpandRate=*/12,
+    /*currentPreemptiveRate=*/345,
+    /*currentAccelerateRate =*/678,
+    /*currentSecondaryDecodedRate=*/901,
+    /*currentSecondaryDiscardedRate=*/0,
+    /*meanWaitingTimeMs=*/-1,
+    /*maxWaitingTimeMs=*/-1,
+    /*packetBufferFlushes=*/0,
+    /*delayedPacketOutageSamples=*/0,
+    /*relativePacketArrivalDelayMs=*/135,
+    /*interruptionCount=*/-1,
+    /*totalInterruptionDurationMs=*/-1};
+const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
+
+struct ConfigHelper {
+  explicit ConfigHelper(bool use_null_audio_processing)
+      : ConfigHelper(rtc::make_ref_counted<MockAudioMixer>(),
+                     use_null_audio_processing) {}
+
+  ConfigHelper(rtc::scoped_refptr<MockAudioMixer> audio_mixer,
+               bool use_null_audio_processing)
+      : audio_mixer_(audio_mixer) {
+    using ::testing::Invoke;
+
+    AudioState::Config config;
+    config.audio_mixer = audio_mixer_;
+    config.audio_processing =
+        use_null_audio_processing
+            ? nullptr
+            : rtc::make_ref_counted<NiceMock<MockAudioProcessing>>();
+    config.audio_device_module =
+        rtc::make_ref_counted<testing::NiceMock<MockAudioDeviceModule>>();
+    audio_state_ = AudioState::Create(config);
+
+    channel_receive_ = new ::testing::StrictMock<MockChannelReceive>();
+    EXPECT_CALL(*channel_receive_, SetNACKStatus(true, 15)).Times(1);
+    EXPECT_CALL(*channel_receive_,
+                RegisterReceiverCongestionControlObjects(&packet_router_))
+        .Times(1);
+    EXPECT_CALL(*channel_receive_, ResetReceiverCongestionControlObjects())
+        .Times(1);
+    EXPECT_CALL(*channel_receive_, SetAssociatedSendChannel(nullptr)).Times(1);
+    EXPECT_CALL(*channel_receive_, SetReceiveCodecs(_))
+        .WillRepeatedly(Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
+          EXPECT_THAT(codecs, ::testing::IsEmpty());
+        }));
+    EXPECT_CALL(*channel_receive_, SetSourceTracker(_));
+    EXPECT_CALL(*channel_receive_, GetLocalSsrc())
+        .WillRepeatedly(Return(kLocalSsrc));
+
+    stream_config_.rtp.local_ssrc = kLocalSsrc;
+    stream_config_.rtp.remote_ssrc = kRemoteSsrc;
+    stream_config_.rtp.nack.rtp_history_ms = 300;
+    stream_config_.rtcp_send_transport = &rtcp_send_transport_;
+    stream_config_.decoder_factory =
+        rtc::make_ref_counted<MockAudioDecoderFactory>();
+  }
+
+  std::unique_ptr<AudioReceiveStreamImpl> CreateAudioReceiveStream() {
+    auto ret = std::make_unique<AudioReceiveStreamImpl>(
+        Clock::GetRealTimeClock(), &packet_router_, stream_config_,
+        audio_state_, &event_log_,
+        std::unique_ptr<voe::ChannelReceiveInterface>(channel_receive_));
+    ret->RegisterWithTransport(&rtp_stream_receiver_controller_);
+    return ret;
+  }
+
+  AudioReceiveStreamInterface::Config& config() { return stream_config_; }
+  rtc::scoped_refptr<MockAudioMixer> audio_mixer() { return audio_mixer_; }
+  MockChannelReceive* channel_receive() { return channel_receive_; }
+
+  void SetupMockForGetStats() {
+    using ::testing::DoAll;
+    using ::testing::SetArgPointee;
+
+    ASSERT_TRUE(channel_receive_);
+    EXPECT_CALL(*channel_receive_, GetRTCPStatistics())
+        .WillOnce(Return(kCallStats));
+    EXPECT_CALL(*channel_receive_, GetDelayEstimate())
+        .WillOnce(Return(kJitterBufferDelay + kPlayoutBufferDelay));
+    EXPECT_CALL(*channel_receive_, GetSpeechOutputLevelFullRange())
+        .WillOnce(Return(kSpeechOutputLevel));
+    EXPECT_CALL(*channel_receive_, GetTotalOutputEnergy())
+        .WillOnce(Return(kTotalOutputEnergy));
+    EXPECT_CALL(*channel_receive_, GetTotalOutputDuration())
+        .WillOnce(Return(kTotalOutputDuration));
+    EXPECT_CALL(*channel_receive_, GetNetworkStatistics(_))
+        .WillOnce(Return(kNetworkStats));
+    EXPECT_CALL(*channel_receive_, GetDecodingCallStatistics())
+        .WillOnce(Return(kAudioDecodeStats));
+    EXPECT_CALL(*channel_receive_, GetReceiveCodec())
+        .WillOnce(Return(kReceiveCodec));
+    EXPECT_CALL(*channel_receive_, GetCurrentEstimatedPlayoutNtpTimestampMs(_))
+        .WillOnce(Return(kPlayoutNtpTimestampMs));
+  }
+
+ private:
+  PacketRouter packet_router_;
+  MockRtcEventLog event_log_;
+  rtc::scoped_refptr<AudioState> audio_state_;
+  rtc::scoped_refptr<MockAudioMixer> audio_mixer_;
+  AudioReceiveStreamInterface::Config stream_config_;
+  ::testing::StrictMock<MockChannelReceive>* channel_receive_ = nullptr;
+  RtpStreamReceiverController rtp_stream_receiver_controller_;
+  MockTransport rtcp_send_transport_;
+};
+
+const std::vector<uint8_t> CreateRtcpSenderReport() {
+  std::vector<uint8_t> packet;
+  const size_t kRtcpSrLength = 28;  // In bytes.
+  packet.resize(kRtcpSrLength);
+  packet[0] = 0x80;  // Version 2.
+  packet[1] = 0xc8;  // PT = 200, SR.
+  // Length in number of 32-bit words - 1.
+  ByteWriter<uint16_t>::WriteBigEndian(&packet[2], 6);
+  ByteWriter<uint32_t>::WriteBigEndian(&packet[4], kLocalSsrc);
+  return packet;
+}
+}  // namespace
+
+TEST(AudioReceiveStreamTest, ConfigToString) {
+  AudioReceiveStreamInterface::Config config;
+  config.rtp.remote_ssrc = kRemoteSsrc;
+  config.rtp.local_ssrc = kLocalSsrc;
+  EXPECT_EQ(
+      "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, nack: "
+      "{rtp_history_ms: 0}}, "
+      "rtcp_send_transport: null}",
+      config.ToString());
+}
+
+TEST(AudioReceiveStreamTest, ConstructDestruct) {
+  test::RunLoop loop;
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, ReceiveRtcpPacket) {
+  test::RunLoop loop;
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+    std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
+    EXPECT_CALL(*helper.channel_receive(),
+                ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size()))
+        .WillOnce(Return());
+    recv_stream->DeliverRtcp(&rtcp_packet[0], rtcp_packet.size());
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, GetStats) {
+  test::RunLoop loop;
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+    helper.SetupMockForGetStats();
+    AudioReceiveStreamInterface::Stats stats =
+        recv_stream->GetStats(/*get_and_clear_legacy_stats=*/true);
+    EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
+    EXPECT_EQ(kCallStats.payload_bytes_received, stats.payload_bytes_received);
+    EXPECT_EQ(kCallStats.header_and_padding_bytes_received,
+              stats.header_and_padding_bytes_received);
+    EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
+              stats.packets_received);
+    EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
+    EXPECT_EQ(kReceiveCodec.second.name, stats.codec_name);
+    EXPECT_EQ(
+        kCallStats.jitterSamples / (kReceiveCodec.second.clockrate_hz / 1000),
+        stats.jitter_ms);
+    EXPECT_EQ(kNetworkStats.currentBufferSize, stats.jitter_buffer_ms);
+    EXPECT_EQ(kNetworkStats.preferredBufferSize,
+              stats.jitter_buffer_preferred_ms);
+    EXPECT_EQ(static_cast<uint32_t>(kJitterBufferDelay + kPlayoutBufferDelay),
+              stats.delay_estimate_ms);
+    EXPECT_EQ(static_cast<int32_t>(kSpeechOutputLevel), stats.audio_level);
+    EXPECT_EQ(kTotalOutputEnergy, stats.total_output_energy);
+    EXPECT_EQ(kNetworkStats.totalSamplesReceived, stats.total_samples_received);
+    EXPECT_EQ(kTotalOutputDuration, stats.total_output_duration);
+    EXPECT_EQ(kNetworkStats.concealedSamples, stats.concealed_samples);
+    EXPECT_EQ(kNetworkStats.concealmentEvents, stats.concealment_events);
+    EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferDelayMs) /
+                  static_cast<double>(rtc::kNumMillisecsPerSec),
+              stats.jitter_buffer_delay_seconds);
+    EXPECT_EQ(kNetworkStats.jitterBufferEmittedCount,
+              stats.jitter_buffer_emitted_count);
+    EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferTargetDelayMs) /
+                  static_cast<double>(rtc::kNumMillisecsPerSec),
+              stats.jitter_buffer_target_delay_seconds);
+    EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferMinimumDelayMs) /
+                  static_cast<double>(rtc::kNumMillisecsPerSec),
+              stats.jitter_buffer_minimum_delay_seconds);
+    EXPECT_EQ(kNetworkStats.insertedSamplesForDeceleration,
+              stats.inserted_samples_for_deceleration);
+    EXPECT_EQ(kNetworkStats.removedSamplesForAcceleration,
+              stats.removed_samples_for_acceleration);
+    EXPECT_EQ(kNetworkStats.fecPacketsReceived, stats.fec_packets_received);
+    EXPECT_EQ(kNetworkStats.fecPacketsDiscarded, stats.fec_packets_discarded);
+    EXPECT_EQ(kNetworkStats.packetsDiscarded, stats.packets_discarded);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate),
+              stats.speech_expand_rate);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDecodedRate),
+              stats.secondary_decoded_rate);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDiscardedRate),
+              stats.secondary_discarded_rate);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentAccelerateRate),
+              stats.accelerate_rate);
+    EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate),
+              stats.preemptive_expand_rate);
+    EXPECT_EQ(kNetworkStats.packetBufferFlushes, stats.jitter_buffer_flushes);
+    EXPECT_EQ(kNetworkStats.delayedPacketOutageSamples,
+              stats.delayed_packet_outage_samples);
+    EXPECT_EQ(static_cast<double>(kNetworkStats.relativePacketArrivalDelayMs) /
+                  static_cast<double>(rtc::kNumMillisecsPerSec),
+              stats.relative_packet_arrival_delay_seconds);
+    EXPECT_EQ(kNetworkStats.interruptionCount, stats.interruption_count);
+    EXPECT_EQ(kNetworkStats.totalInterruptionDurationMs,
+              stats.total_interruption_duration_ms);
+
+    EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator,
+              stats.decoding_calls_to_silence_generator);
+    EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
+    EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
+    EXPECT_EQ(kAudioDecodeStats.decoded_neteq_plc, stats.decoding_plc);
+    EXPECT_EQ(kAudioDecodeStats.decoded_codec_plc, stats.decoding_codec_plc);
+    EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
+    EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
+    EXPECT_EQ(kAudioDecodeStats.decoded_muted_output,
+              stats.decoding_muted_output);
+    EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
+              stats.capture_start_ntp_time_ms);
+    EXPECT_EQ(kPlayoutNtpTimestampMs, stats.estimated_playout_ntp_timestamp_ms);
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, SetGain) {
+  test::RunLoop loop;
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+    EXPECT_CALL(*helper.channel_receive(),
+                SetChannelOutputVolumeScaling(FloatEq(0.765f)));
+    recv_stream->SetGain(0.765f);
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, StreamsShouldBeAddedToMixerOnceOnStart) {
+  test::RunLoop loop;
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper1(use_null_audio_processing);
+    ConfigHelper helper2(helper1.audio_mixer(), use_null_audio_processing);
+    auto recv_stream1 = helper1.CreateAudioReceiveStream();
+    auto recv_stream2 = helper2.CreateAudioReceiveStream();
+
+    EXPECT_CALL(*helper1.channel_receive(), StartPlayout()).Times(1);
+    EXPECT_CALL(*helper2.channel_receive(), StartPlayout()).Times(1);
+    EXPECT_CALL(*helper1.channel_receive(), StopPlayout()).Times(1);
+    EXPECT_CALL(*helper2.channel_receive(), StopPlayout()).Times(1);
+    EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream1.get()))
+        .WillOnce(Return(true));
+    EXPECT_CALL(*helper1.audio_mixer(), AddSource(recv_stream2.get()))
+        .WillOnce(Return(true));
+    EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream1.get()))
+        .Times(1);
+    EXPECT_CALL(*helper1.audio_mixer(), RemoveSource(recv_stream2.get()))
+        .Times(1);
+
+    recv_stream1->Start();
+    recv_stream2->Start();
+
+    // One more should not result in any more mixer sources added.
+    recv_stream1->Start();
+
+    // Stop stream before it is being destructed.
+    recv_stream2->Stop();
+
+    recv_stream1->UnregisterFromTransport();
+    recv_stream2->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
+  test::RunLoop loop;
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+
+    auto new_config = helper.config();
+
+    MockChannelReceive& channel_receive = *helper.channel_receive();
+
+    // TODO(tommi, nisse): This applies new extensions to the internal config,
+    // but there's nothing that actually verifies that the changes take effect.
+    // In fact Call manages the extensions separately in Call::ReceiveRtpConfig
+    // and changing this config value (there seem to be a few copies), doesn't
+    // affect that logic.
+    recv_stream->ReconfigureForTesting(new_config);
+
+    new_config.decoder_map.emplace(1, SdpAudioFormat("foo", 8000, 1));
+    EXPECT_CALL(channel_receive, SetReceiveCodecs(new_config.decoder_map));
+    recv_stream->SetDecoderMap(new_config.decoder_map);
+
+    EXPECT_CALL(channel_receive, SetNACKStatus(true, 15 + 1)).Times(1);
+    recv_stream->SetNackHistory(300 + 20);
+
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+TEST(AudioReceiveStreamTest, ReconfigureWithFrameDecryptor) {
+  test::RunLoop loop;
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(use_null_audio_processing);
+    auto recv_stream = helper.CreateAudioReceiveStream();
+
+    auto new_config_0 = helper.config();
+    rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_0(
+        rtc::make_ref_counted<MockFrameDecryptor>());
+    new_config_0.frame_decryptor = mock_frame_decryptor_0;
+
+    // TODO(tommi): While this changes the internal config value, it doesn't
+    // actually change what frame_decryptor is used. WebRtcAudioReceiveStream
+    // recreates the whole instance in order to change this value.
+    // So, it's not clear if changing this post initialization needs to be
+    // supported.
+    recv_stream->ReconfigureForTesting(new_config_0);
+
+    auto new_config_1 = helper.config();
+    rtc::scoped_refptr<FrameDecryptorInterface> mock_frame_decryptor_1(
+        rtc::make_ref_counted<MockFrameDecryptor>());
+    new_config_1.frame_decryptor = mock_frame_decryptor_1;
+    new_config_1.crypto_options.sframe.require_frame_encryption = true;
+    recv_stream->ReconfigureForTesting(new_config_1);
+    recv_stream->UnregisterFromTransport();
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/audio_send_stream.cc b/third_party/libwebrtc/audio/audio_send_stream.cc
new file mode 100644
index 0000000000..bffb910832
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_send_stream.cc
@@ -0,0 +1,921 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_send_stream.h"
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/call/transport.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/function_view.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/task_queue/task_queue_base.h"
+#include "audio/audio_state.h"
+#include "audio/channel_send.h"
+#include "audio/conversion.h"
+#include "call/rtp_config.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "common_audio/vad/include/vad.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
+#include "logging/rtc_event_log/rtc_stream_config.h"
+#include "media/base/media_channel.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/audio_format_to_string.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+namespace {
+
+void UpdateEventLogStreamConfig(RtcEventLog* event_log,
+                                const AudioSendStream::Config& config,
+                                const AudioSendStream::Config* old_config) {
+  using SendCodecSpec = AudioSendStream::Config::SendCodecSpec;
+  // Only update if any of the things we log have changed.
+  auto payload_types_equal = [](const absl::optional<SendCodecSpec>& a,
+                                const absl::optional<SendCodecSpec>& b) {
+    if (a.has_value() && b.has_value()) {
+      return a->format.name == b->format.name &&
+             a->payload_type == b->payload_type;
+    }
+    return !a.has_value() && !b.has_value();
+  };
+
+  if (old_config && config.rtp.ssrc == old_config->rtp.ssrc &&
+      config.rtp.extensions == old_config->rtp.extensions &&
+      payload_types_equal(config.send_codec_spec,
+                          old_config->send_codec_spec)) {
+    return;
+  }
+
+  auto rtclog_config = std::make_unique<rtclog::StreamConfig>();
+  rtclog_config->local_ssrc = config.rtp.ssrc;
+  rtclog_config->rtp_extensions = config.rtp.extensions;
+  if (config.send_codec_spec) {
+    rtclog_config->codecs.emplace_back(config.send_codec_spec->format.name,
+                                       config.send_codec_spec->payload_type, 0);
+  }
+  event_log->Log(std::make_unique<RtcEventAudioSendStreamConfig>(
+      std::move(rtclog_config)));
+}
+
+}  // namespace
+
+constexpr char AudioAllocationConfig::kKey[];
+
+std::unique_ptr<StructParametersParser> AudioAllocationConfig::Parser() {
+  return StructParametersParser::Create(       //
+      "min", &min_bitrate,                     //
+      "max", &max_bitrate,                     //
+      "prio_rate", &priority_bitrate,          //
+      "prio_rate_raw", &priority_bitrate_raw,  //
+      "rate_prio", &bitrate_priority);
+}
+
+AudioAllocationConfig::AudioAllocationConfig(
+    const FieldTrialsView& field_trials) {
+  Parser()->Parse(field_trials.Lookup(kKey));
+  if (priority_bitrate_raw && !priority_bitrate.IsZero()) {
+    RTC_LOG(LS_WARNING) << "'priority_bitrate' and '_raw' are mutually "
+                           "exclusive but both were configured.";
+  }
+}
+
+namespace internal {
+AudioSendStream::AudioSendStream(
+    Clock* clock,
+    const webrtc::AudioSendStream::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    TaskQueueFactory* task_queue_factory,
+    RtpTransportControllerSendInterface* rtp_transport,
+    BitrateAllocatorInterface* bitrate_allocator,
+    RtcEventLog* event_log,
+    RtcpRttStats* rtcp_rtt_stats,
+    const absl::optional<RtpState>& suspended_rtp_state,
+    const FieldTrialsView& field_trials)
+    : AudioSendStream(clock,
+                      config,
+                      audio_state,
+                      task_queue_factory,
+                      rtp_transport,
+                      bitrate_allocator,
+                      event_log,
+                      suspended_rtp_state,
+                      voe::CreateChannelSend(clock,
+                                             task_queue_factory,
+                                             config.send_transport,
+                                             rtcp_rtt_stats,
+                                             event_log,
+                                             config.frame_encryptor.get(),
+                                             config.crypto_options,
+                                             config.rtp.extmap_allow_mixed,
+                                             config.rtcp_report_interval_ms,
+                                             config.rtp.ssrc,
+                                             config.frame_transformer,
+                                             rtp_transport,
+                                             field_trials),
+                      field_trials) {}
+
+AudioSendStream::AudioSendStream(
+    Clock* clock,
+    const webrtc::AudioSendStream::Config& config,
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    TaskQueueFactory* task_queue_factory,
+    RtpTransportControllerSendInterface* rtp_transport,
+    BitrateAllocatorInterface* bitrate_allocator,
+    RtcEventLog* event_log,
+    const absl::optional<RtpState>& suspended_rtp_state,
+    std::unique_ptr<voe::ChannelSendInterface> channel_send,
+    const FieldTrialsView& field_trials)
+    : clock_(clock),
+      field_trials_(field_trials),
+      allocate_audio_without_feedback_(
+          field_trials_.IsEnabled("WebRTC-Audio-ABWENoTWCC")),
+      enable_audio_alr_probing_(
+          !field_trials_.IsDisabled("WebRTC-Audio-AlrProbing")),
+      allocation_settings_(field_trials_),
+      config_(Config(/*send_transport=*/nullptr)),
+      audio_state_(audio_state),
+      channel_send_(std::move(channel_send)),
+      event_log_(event_log),
+      use_legacy_overhead_calculation_(
+          field_trials_.IsEnabled("WebRTC-Audio-LegacyOverhead")),
+      bitrate_allocator_(bitrate_allocator),
+      rtp_transport_(rtp_transport),
+      rtp_rtcp_module_(channel_send_->GetRtpRtcp()),
+      suspended_rtp_state_(suspended_rtp_state) {
+  RTC_LOG(LS_INFO) << "AudioSendStream: " << config.rtp.ssrc;
+  RTC_DCHECK(audio_state_);
+  RTC_DCHECK(channel_send_);
+  RTC_DCHECK(bitrate_allocator_);
+  RTC_DCHECK(rtp_transport);
+
+  RTC_DCHECK(rtp_rtcp_module_);
+
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  ConfigureStream(config, true, nullptr);
+  UpdateCachedTargetAudioBitrateConstraints();
+}
+
+AudioSendStream::~AudioSendStream() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.rtp.ssrc;
+  RTC_DCHECK(!sending_);
+  channel_send_->ResetSenderCongestionControlObjects();
+}
+
+const webrtc::AudioSendStream::Config& AudioSendStream::GetConfig() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return config_;
+}
+
+void AudioSendStream::Reconfigure(
+    const webrtc::AudioSendStream::Config& new_config,
+    SetParametersCallback callback) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  ConfigureStream(new_config, false, std::move(callback));
+}
+
+AudioSendStream::ExtensionIds AudioSendStream::FindExtensionIds(
+    const std::vector<RtpExtension>& extensions) {
+  ExtensionIds ids;
+  for (const auto& extension : extensions) {
+    if (extension.uri == RtpExtension::kAudioLevelUri) {
+      ids.audio_level = extension.id;
+    } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
+      ids.abs_send_time = extension.id;
+    } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
+      ids.transport_sequence_number = extension.id;
+    } else if (extension.uri == RtpExtension::kMidUri) {
+      ids.mid = extension.id;
+    } else if (extension.uri == RtpExtension::kRidUri) {
+      ids.rid = extension.id;
+    } else if (extension.uri == RtpExtension::kRepairedRidUri) {
+      ids.repaired_rid = extension.id;
+    } else if (extension.uri == RtpExtension::kAbsoluteCaptureTimeUri) {
+      ids.abs_capture_time = extension.id;
+    }
+  }
+  return ids;
+}
+
+int AudioSendStream::TransportSeqNumId(const AudioSendStream::Config& config) {
+  return FindExtensionIds(config.rtp.extensions).transport_sequence_number;
+}
+
+void AudioSendStream::ConfigureStream(
+    const webrtc::AudioSendStream::Config& new_config,
+    bool first_time,
+    SetParametersCallback callback) {
+  RTC_LOG(LS_INFO) << "AudioSendStream::ConfigureStream: "
+                   << new_config.ToString();
+  UpdateEventLogStreamConfig(event_log_, new_config,
+                             first_time ? nullptr : &config_);
+
+  const auto& old_config = config_;
+
+  // Configuration parameters which cannot be changed.
+  RTC_DCHECK(first_time ||
+             old_config.send_transport == new_config.send_transport);
+  RTC_DCHECK(first_time || old_config.rtp.ssrc == new_config.rtp.ssrc);
+  if (suspended_rtp_state_ && first_time) {
+    rtp_rtcp_module_->SetRtpState(*suspended_rtp_state_);
+  }
+  if (first_time || old_config.rtp.c_name != new_config.rtp.c_name) {
+    channel_send_->SetRTCP_CNAME(new_config.rtp.c_name);
+  }
+
+  // Enable the frame encryptor if a new frame encryptor has been provided.
+  if (first_time || new_config.frame_encryptor != old_config.frame_encryptor) {
+    channel_send_->SetFrameEncryptor(new_config.frame_encryptor);
+  }
+
+  if (first_time ||
+      new_config.frame_transformer != old_config.frame_transformer) {
+    channel_send_->SetEncoderToPacketizerFrameTransformer(
+        new_config.frame_transformer);
+  }
+
+  if (first_time ||
+      new_config.rtp.extmap_allow_mixed != old_config.rtp.extmap_allow_mixed) {
+    rtp_rtcp_module_->SetExtmapAllowMixed(new_config.rtp.extmap_allow_mixed);
+  }
+
+  const ExtensionIds old_ids = FindExtensionIds(old_config.rtp.extensions);
+  const ExtensionIds new_ids = FindExtensionIds(new_config.rtp.extensions);
+
+  // Audio level indication
+  if (first_time || new_ids.audio_level != old_ids.audio_level) {
+    channel_send_->SetSendAudioLevelIndicationStatus(new_ids.audio_level != 0,
+                                                     new_ids.audio_level);
+  }
+
+  if (first_time || new_ids.abs_send_time != old_ids.abs_send_time) {
+    absl::string_view uri = AbsoluteSendTime::Uri();
+    rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(uri);
+    if (new_ids.abs_send_time) {
+      rtp_rtcp_module_->RegisterRtpHeaderExtension(uri, new_ids.abs_send_time);
+    }
+  }
+
+  bool transport_seq_num_id_changed =
+      new_ids.transport_sequence_number != old_ids.transport_sequence_number;
+  if (first_time ||
+      (transport_seq_num_id_changed && !allocate_audio_without_feedback_)) {
+    if (!first_time) {
+      channel_send_->ResetSenderCongestionControlObjects();
+    }
+
+    if (!allocate_audio_without_feedback_ &&
+        new_ids.transport_sequence_number != 0) {
+      rtp_rtcp_module_->RegisterRtpHeaderExtension(
+          TransportSequenceNumber::Uri(), new_ids.transport_sequence_number);
+      // Probing in application limited region is only used in combination with
+      // send side congestion control, wich depends on feedback packets which
+      // requires transport sequence numbers to be enabled.
+      // Optionally request ALR probing but do not override any existing
+      // request from other streams.
+      if (enable_audio_alr_probing_) {
+        rtp_transport_->EnablePeriodicAlrProbing(true);
+      }
+    }
+    channel_send_->RegisterSenderCongestionControlObjects(rtp_transport_);
+  }
+  // MID RTP header extension.
+  if ((first_time || new_ids.mid != old_ids.mid ||
+       new_config.rtp.mid != old_config.rtp.mid) &&
+      new_ids.mid != 0 && !new_config.rtp.mid.empty()) {
+    rtp_rtcp_module_->RegisterRtpHeaderExtension(RtpMid::Uri(), new_ids.mid);
+    rtp_rtcp_module_->SetMid(new_config.rtp.mid);
+  }
+
+  if (first_time || new_ids.abs_capture_time != old_ids.abs_capture_time) {
+    absl::string_view uri = AbsoluteCaptureTimeExtension::Uri();
+    rtp_rtcp_module_->DeregisterSendRtpHeaderExtension(uri);
+    if (new_ids.abs_capture_time) {
+      rtp_rtcp_module_->RegisterRtpHeaderExtension(uri,
+                                                   new_ids.abs_capture_time);
+    }
+  }
+
+  if (!ReconfigureSendCodec(new_config)) {
+    RTC_LOG(LS_ERROR) << "Failed to set up send codec state.";
+
+    webrtc::InvokeSetParametersCallback(
+        callback, webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR,
+                                   "Failed to set up send codec state."));
+  }
+
+  // Set currently known overhead (used in ANA, opus only).
+  {
+    MutexLock lock(&overhead_per_packet_lock_);
+    UpdateOverheadForEncoder();
+  }
+
+  channel_send_->CallEncoder([this](AudioEncoder* encoder) {
+    RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+    if (!encoder) {
+      return;
+    }
+    frame_length_range_ = encoder->GetFrameLengthRange();
+    UpdateCachedTargetAudioBitrateConstraints();
+  });
+
+  if (sending_) {
+    ReconfigureBitrateObserver(new_config);
+  }
+
+  config_ = new_config;
+  if (!first_time) {
+    UpdateCachedTargetAudioBitrateConstraints();
+  }
+
+  webrtc::InvokeSetParametersCallback(callback, webrtc::RTCError::OK());
+}
+
+void AudioSendStream::Start() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (sending_) {
+    return;
+  }
+  if (!config_.has_dscp && config_.min_bitrate_bps != -1 &&
+      config_.max_bitrate_bps != -1 &&
+      (allocate_audio_without_feedback_ || TransportSeqNumId(config_) != 0)) {
+    rtp_transport_->AccountForAudioPacketsInPacedSender(true);
+    rtp_transport_->IncludeOverheadInPacedSender();
+    rtp_rtcp_module_->SetAsPartOfAllocation(true);
+    ConfigureBitrateObserver();
+  } else {
+    rtp_rtcp_module_->SetAsPartOfAllocation(false);
+  }
+  channel_send_->StartSend();
+  sending_ = true;
+  audio_state()->AddSendingStream(this, encoder_sample_rate_hz_,
+                                  encoder_num_channels_);
+}
+
+void AudioSendStream::Stop() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!sending_) {
+    return;
+  }
+
+  RemoveBitrateObserver();
+  channel_send_->StopSend();
+  sending_ = false;
+  audio_state()->RemoveSendingStream(this);
+}
+
+void AudioSendStream::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
+  RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
+  RTC_DCHECK_GT(audio_frame->sample_rate_hz_, 0);
+  TRACE_EVENT0("webrtc", "AudioSendStream::SendAudioData");
+  double duration = static_cast<double>(audio_frame->samples_per_channel_) /
+                    audio_frame->sample_rate_hz_;
+  {
+    // Note: SendAudioData() passes the frame further down the pipeline and it
+    // may eventually get sent. But this method is invoked even if we are not
+    // connected, as long as we have an AudioSendStream (created as a result of
+    // an O/A exchange). This means that we are calculating audio levels whether
+    // or not we are sending samples.
+    // TODO(https://crbug.com/webrtc/10771): All "media-source" related stats
+    // should move from send-streams to the local audio sources or tracks; a
+    // send-stream should not be required to read the microphone audio levels.
+    MutexLock lock(&audio_level_lock_);
+    audio_level_.ComputeLevel(*audio_frame, duration);
+  }
+  channel_send_->ProcessAndEncodeAudio(std::move(audio_frame));
+}
+
+bool AudioSendStream::SendTelephoneEvent(int payload_type,
+                                         int payload_frequency,
+                                         int event,
+                                         int duration_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_send_->SetSendTelephoneEventPayloadType(payload_type,
+                                                  payload_frequency);
+  return channel_send_->SendTelephoneEventOutband(event, duration_ms);
+}
+
+void AudioSendStream::SetMuted(bool muted) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_send_->SetInputMute(muted);
+}
+
+webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
+  return GetStats(true);
+}
+
+webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
+    bool has_remote_tracks) const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  webrtc::AudioSendStream::Stats stats;
+  stats.local_ssrc = config_.rtp.ssrc;
+  stats.target_bitrate_bps = channel_send_->GetTargetBitrate();
+
+  webrtc::CallSendStatistics call_stats = channel_send_->GetRTCPStatistics();
+  stats.rtcp_packet_type_counts = call_stats.rtcp_packet_type_counts;
+  stats.payload_bytes_sent = call_stats.payload_bytes_sent;
+  stats.header_and_padding_bytes_sent =
+      call_stats.header_and_padding_bytes_sent;
+  stats.retransmitted_bytes_sent = call_stats.retransmitted_bytes_sent;
+  stats.packets_sent = call_stats.packetsSent;
+  stats.total_packet_send_delay = call_stats.total_packet_send_delay;
+  stats.retransmitted_packets_sent = call_stats.retransmitted_packets_sent;
+  // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
+  // returns 0 to indicate an error value.
+  if (call_stats.rttMs > 0) {
+    stats.rtt_ms = call_stats.rttMs;
+  }
+  if (config_.send_codec_spec) {
+    const auto& spec = *config_.send_codec_spec;
+    stats.codec_name = spec.format.name;
+    stats.codec_payload_type = spec.payload_type;
+
+    // Get data from the last remote RTCP report.
+    for (const ReportBlockData& block :
+         channel_send_->GetRemoteRTCPReportBlocks()) {
+      // Lookup report for send ssrc only.
+      if (block.source_ssrc() == stats.local_ssrc) {
+        stats.packets_lost = block.cumulative_lost();
+        stats.fraction_lost = block.fraction_lost();
+        if (spec.format.clockrate_hz > 0) {
+          stats.jitter_ms = block.jitter(spec.format.clockrate_hz).ms();
+        }
+        break;
+      }
+    }
+  }
+
+  {
+    MutexLock lock(&audio_level_lock_);
+    stats.audio_level = audio_level_.LevelFullRange();
+    stats.total_input_energy = audio_level_.TotalEnergy();
+    stats.total_input_duration = audio_level_.TotalDuration();
+  }
+
+  stats.ana_statistics = channel_send_->GetANAStatistics();
+
+  AudioProcessing* ap = audio_state_->audio_processing();
+  if (ap) {
+    stats.apm_statistics = ap->GetStatistics(has_remote_tracks);
+  }
+
+  stats.report_block_datas = std::move(call_stats.report_block_datas);
+
+  stats.nacks_received = call_stats.nacks_received;
+
+  return stats;
+}
+
+void AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  channel_send_->ReceivedRTCPPacket(packet, length);
+
+  {
+    // Poll if overhead has changed, which it can do if ack triggers us to stop
+    // sending mid/rid.
+    MutexLock lock(&overhead_per_packet_lock_);
+    UpdateOverheadForEncoder();
+  }
+  UpdateCachedTargetAudioBitrateConstraints();
+}
+
+uint32_t AudioSendStream::OnBitrateUpdated(BitrateAllocationUpdate update) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+
+  // Pick a target bitrate between the constraints. Overrules the allocator if
+  // it 1) allocated a bitrate of zero to disable the stream or 2) allocated a
+  // higher than max to allow for e.g. extra FEC.
+  RTC_DCHECK(cached_constraints_.has_value());
+  update.target_bitrate.Clamp(cached_constraints_->min,
+                              cached_constraints_->max);
+  update.stable_target_bitrate.Clamp(cached_constraints_->min,
+                                     cached_constraints_->max);
+
+  channel_send_->OnBitrateAllocation(update);
+
+  // The amount of audio protection is not exposed by the encoder, hence
+  // always returning 0.
+  return 0;
+}
+
+void AudioSendStream::SetTransportOverhead(
+    int transport_overhead_per_packet_bytes) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  {
+    MutexLock lock(&overhead_per_packet_lock_);
+    transport_overhead_per_packet_bytes_ = transport_overhead_per_packet_bytes;
+    UpdateOverheadForEncoder();
+  }
+  UpdateCachedTargetAudioBitrateConstraints();
+}
+
+void AudioSendStream::UpdateOverheadForEncoder() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  size_t overhead_per_packet_bytes = GetPerPacketOverheadBytes();
+  if (overhead_per_packet_ == overhead_per_packet_bytes) {
+    return;
+  }
+  overhead_per_packet_ = overhead_per_packet_bytes;
+
+  channel_send_->CallEncoder([&](AudioEncoder* encoder) {
+    encoder->OnReceivedOverhead(overhead_per_packet_bytes);
+  });
+  if (total_packet_overhead_bytes_ != overhead_per_packet_bytes) {
+    total_packet_overhead_bytes_ = overhead_per_packet_bytes;
+    if (registered_with_allocator_) {
+      ConfigureBitrateObserver();
+    }
+  }
+}
+
+size_t AudioSendStream::TestOnlyGetPerPacketOverheadBytes() const {
+  MutexLock lock(&overhead_per_packet_lock_);
+  return GetPerPacketOverheadBytes();
+}
+
+size_t AudioSendStream::GetPerPacketOverheadBytes() const {
+  return transport_overhead_per_packet_bytes_ +
+         rtp_rtcp_module_->ExpectedPerPacketOverhead();
+}
+
+RtpState AudioSendStream::GetRtpState() const {
+  return rtp_rtcp_module_->GetRtpState();
+}
+
+const voe::ChannelSendInterface* AudioSendStream::GetChannel() const {
+  return channel_send_.get();
+}
+
+internal::AudioState* AudioSendStream::audio_state() {
+  internal::AudioState* audio_state =
+      static_cast<internal::AudioState*>(audio_state_.get());
+  RTC_DCHECK(audio_state);
+  return audio_state;
+}
+
+const internal::AudioState* AudioSendStream::audio_state() const {
+  internal::AudioState* audio_state =
+      static_cast<internal::AudioState*>(audio_state_.get());
+  RTC_DCHECK(audio_state);
+  return audio_state;
+}
+
+void AudioSendStream::StoreEncoderProperties(int sample_rate_hz,
+                                             size_t num_channels) {
+  encoder_sample_rate_hz_ = sample_rate_hz;
+  encoder_num_channels_ = num_channels;
+  if (sending_) {
+    // Update AudioState's information about the stream.
+    audio_state()->AddSendingStream(this, sample_rate_hz, num_channels);
+  }
+}
+
+// Apply current codec settings to a single voe::Channel used for sending.
+bool AudioSendStream::SetupSendCodec(const Config& new_config) {
+  RTC_DCHECK(new_config.send_codec_spec);
+  const auto& spec = *new_config.send_codec_spec;
+
+  RTC_DCHECK(new_config.encoder_factory);
+  std::unique_ptr<AudioEncoder> encoder =
+      new_config.encoder_factory->MakeAudioEncoder(
+          spec.payload_type, spec.format, new_config.codec_pair_id);
+
+  if (!encoder) {
+    RTC_DLOG(LS_ERROR) << "Unable to create encoder for "
+                       << rtc::ToString(spec.format);
+    return false;
+  }
+
+  // If a bitrate has been specified for the codec, use it over the
+  // codec's default.
+  if (spec.target_bitrate_bps) {
+    encoder->OnReceivedTargetAudioBitrate(*spec.target_bitrate_bps);
+  }
+
+  // Enable ANA if configured (currently only used by Opus).
+  if (new_config.audio_network_adaptor_config) {
+    if (encoder->EnableAudioNetworkAdaptor(
+            *new_config.audio_network_adaptor_config, event_log_)) {
+      RTC_LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
+                       << new_config.rtp.ssrc;
+    } else {
+      RTC_LOG(LS_INFO) << "Failed to enable Audio network adaptor on SSRC "
+                       << new_config.rtp.ssrc;
+    }
+  }
+
+  // Wrap the encoder in an AudioEncoderCNG, if VAD is enabled.
+  if (spec.cng_payload_type) {
+    AudioEncoderCngConfig cng_config;
+    cng_config.num_channels = encoder->NumChannels();
+    cng_config.payload_type = *spec.cng_payload_type;
+    cng_config.speech_encoder = std::move(encoder);
+    cng_config.vad_mode = Vad::kVadNormal;
+    encoder = CreateComfortNoiseEncoder(std::move(cng_config));
+
+    RegisterCngPayloadType(*spec.cng_payload_type,
+                           new_config.send_codec_spec->format.clockrate_hz);
+  }
+
+  // Wrap the encoder in a RED encoder, if RED is enabled.
+  if (spec.red_payload_type) {
+    AudioEncoderCopyRed::Config red_config;
+    red_config.payload_type = *spec.red_payload_type;
+    red_config.speech_encoder = std::move(encoder);
+    encoder = std::make_unique<AudioEncoderCopyRed>(std::move(red_config),
+                                                    field_trials_);
+  }
+
+  // Set currently known overhead (used in ANA, opus only).
+  // If overhead changes later, it will be updated in UpdateOverheadForEncoder.
+  {
+    MutexLock lock(&overhead_per_packet_lock_);
+    size_t overhead = GetPerPacketOverheadBytes();
+    if (overhead > 0) {
+      encoder->OnReceivedOverhead(overhead);
+    }
+  }
+
+  StoreEncoderProperties(encoder->SampleRateHz(), encoder->NumChannels());
+  channel_send_->SetEncoder(new_config.send_codec_spec->payload_type,
+                            std::move(encoder));
+
+  return true;
+}
+
+bool AudioSendStream::ReconfigureSendCodec(const Config& new_config) {
+  const auto& old_config = config_;
+
+  if (!new_config.send_codec_spec) {
+    // We cannot de-configure a send codec. So we will do nothing.
+    // By design, the send codec should have not been configured.
+    RTC_DCHECK(!old_config.send_codec_spec);
+    return true;
+  }
+
+  if (new_config.send_codec_spec == old_config.send_codec_spec &&
+      new_config.audio_network_adaptor_config ==
+          old_config.audio_network_adaptor_config) {
+    return true;
+  }
+
+  // If we have no encoder, or the format or payload type's changed, create a
+  // new encoder.
+  if (!old_config.send_codec_spec ||
+      new_config.send_codec_spec->format !=
+          old_config.send_codec_spec->format ||
+      new_config.send_codec_spec->payload_type !=
+          old_config.send_codec_spec->payload_type ||
+      new_config.send_codec_spec->red_payload_type !=
+          old_config.send_codec_spec->red_payload_type) {
+    return SetupSendCodec(new_config);
+  }
+
+  const absl::optional<int>& new_target_bitrate_bps =
+      new_config.send_codec_spec->target_bitrate_bps;
+  // If a bitrate has been specified for the codec, use it over the
+  // codec's default.
+  if (new_target_bitrate_bps &&
+      new_target_bitrate_bps !=
+          old_config.send_codec_spec->target_bitrate_bps) {
+    channel_send_->CallEncoder([&](AudioEncoder* encoder) {
+      encoder->OnReceivedTargetAudioBitrate(*new_target_bitrate_bps);
+    });
+  }
+
+  ReconfigureANA(new_config);
+  ReconfigureCNG(new_config);
+
+  return true;
+}
+
+void AudioSendStream::ReconfigureANA(const Config& new_config) {
+  if (new_config.audio_network_adaptor_config ==
+      config_.audio_network_adaptor_config) {
+    return;
+  }
+  if (new_config.audio_network_adaptor_config) {
+    // This lock needs to be acquired before CallEncoder, since it aquires
+    // another lock and we need to maintain the same order at all call sites to
+    // avoid deadlock.
+    MutexLock lock(&overhead_per_packet_lock_);
+    size_t overhead = GetPerPacketOverheadBytes();
+    channel_send_->CallEncoder([&](AudioEncoder* encoder) {
+      if (encoder->EnableAudioNetworkAdaptor(
+              *new_config.audio_network_adaptor_config, event_log_)) {
+        RTC_LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
+                         << new_config.rtp.ssrc;
+        if (overhead > 0) {
+          encoder->OnReceivedOverhead(overhead);
+        }
+      } else {
+        RTC_LOG(LS_INFO) << "Failed to enable Audio network adaptor on SSRC "
+                         << new_config.rtp.ssrc;
+      }
+    });
+  } else {
+    channel_send_->CallEncoder(
+        [&](AudioEncoder* encoder) { encoder->DisableAudioNetworkAdaptor(); });
+    RTC_LOG(LS_INFO) << "Audio network adaptor disabled on SSRC "
+                     << new_config.rtp.ssrc;
+  }
+}
+
+void AudioSendStream::ReconfigureCNG(const Config& new_config) {
+  if (new_config.send_codec_spec->cng_payload_type ==
+      config_.send_codec_spec->cng_payload_type) {
+    return;
+  }
+
+  // Register the CNG payload type if it's been added, don't do anything if CNG
+  // is removed. Payload types must not be redefined.
+  if (new_config.send_codec_spec->cng_payload_type) {
+    RegisterCngPayloadType(*new_config.send_codec_spec->cng_payload_type,
+                           new_config.send_codec_spec->format.clockrate_hz);
+  }
+
+  // Wrap or unwrap the encoder in an AudioEncoderCNG.
+  channel_send_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
+    std::unique_ptr<AudioEncoder> old_encoder(std::move(*encoder_ptr));
+    auto sub_encoders = old_encoder->ReclaimContainedEncoders();
+    if (!sub_encoders.empty()) {
+      // Replace enc with its sub encoder. We need to put the sub
+      // encoder in a temporary first, since otherwise the old value
+      // of enc would be destroyed before the new value got assigned,
+      // which would be bad since the new value is a part of the old
+      // value.
+      auto tmp = std::move(sub_encoders[0]);
+      old_encoder = std::move(tmp);
+    }
+    if (new_config.send_codec_spec->cng_payload_type) {
+      AudioEncoderCngConfig config;
+      config.speech_encoder = std::move(old_encoder);
+      config.num_channels = config.speech_encoder->NumChannels();
+      config.payload_type = *new_config.send_codec_spec->cng_payload_type;
+      config.vad_mode = Vad::kVadNormal;
+      *encoder_ptr = CreateComfortNoiseEncoder(std::move(config));
+    } else {
+      *encoder_ptr = std::move(old_encoder);
+    }
+  });
+}
+
+void AudioSendStream::ReconfigureBitrateObserver(
+    const webrtc::AudioSendStream::Config& new_config) {
+  // Since the Config's default is for both of these to be -1, this test will
+  // allow us to configure the bitrate observer if the new config has bitrate
+  // limits set, but would only have us call RemoveBitrateObserver if we were
+  // previously configured with bitrate limits.
+  if (config_.min_bitrate_bps == new_config.min_bitrate_bps &&
+      config_.max_bitrate_bps == new_config.max_bitrate_bps &&
+      config_.bitrate_priority == new_config.bitrate_priority &&
+      TransportSeqNumId(config_) == TransportSeqNumId(new_config) &&
+      config_.audio_network_adaptor_config ==
+          new_config.audio_network_adaptor_config) {
+    return;
+  }
+
+  if (!new_config.has_dscp && new_config.min_bitrate_bps != -1 &&
+      new_config.max_bitrate_bps != -1 && TransportSeqNumId(new_config) != 0) {
+    rtp_transport_->AccountForAudioPacketsInPacedSender(true);
+    rtp_transport_->IncludeOverheadInPacedSender();
+    // We may get a callback immediately as the observer is registered, so
+    // make sure the bitrate limits in config_ are up-to-date.
+    config_.min_bitrate_bps = new_config.min_bitrate_bps;
+    config_.max_bitrate_bps = new_config.max_bitrate_bps;
+
+    config_.bitrate_priority = new_config.bitrate_priority;
+    ConfigureBitrateObserver();
+    rtp_rtcp_module_->SetAsPartOfAllocation(true);
+  } else {
+    rtp_transport_->AccountForAudioPacketsInPacedSender(false);
+    RemoveBitrateObserver();
+    rtp_rtcp_module_->SetAsPartOfAllocation(false);
+  }
+}
+
+void AudioSendStream::ConfigureBitrateObserver() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // This either updates the current observer or adds a new observer.
+  // TODO(srte): Add overhead compensation here.
+  auto constraints = GetMinMaxBitrateConstraints();
+  RTC_DCHECK(constraints.has_value());
+
+  DataRate priority_bitrate = allocation_settings_.priority_bitrate;
+  if (use_legacy_overhead_calculation_) {
+    // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
+    constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;
+    const TimeDelta kMinPacketDuration = TimeDelta::Millis(20);
+    DataRate max_overhead =
+        DataSize::Bytes(kOverheadPerPacket) / kMinPacketDuration;
+    priority_bitrate += max_overhead;
+  } else {
+    RTC_DCHECK(frame_length_range_);
+    const DataSize overhead_per_packet =
+        DataSize::Bytes(total_packet_overhead_bytes_);
+    DataRate min_overhead = overhead_per_packet / frame_length_range_->second;
+    priority_bitrate += min_overhead;
+  }
+
+  if (allocation_settings_.priority_bitrate_raw) {
+    priority_bitrate = *allocation_settings_.priority_bitrate_raw;
+  }
+
+  bitrate_allocator_->AddObserver(
+      this,
+      MediaStreamAllocationConfig{
+          constraints->min.bps<uint32_t>(), constraints->max.bps<uint32_t>(), 0,
+          priority_bitrate.bps(), true,
+          allocation_settings_.bitrate_priority.value_or(
+              config_.bitrate_priority)});
+
+  registered_with_allocator_ = true;
+}
+
+void AudioSendStream::RemoveBitrateObserver() {
+  registered_with_allocator_ = false;
+  bitrate_allocator_->RemoveObserver(this);
+}
+
+absl::optional<AudioSendStream::TargetAudioBitrateConstraints>
+AudioSendStream::GetMinMaxBitrateConstraints() const {
+  if (config_.min_bitrate_bps < 0 || config_.max_bitrate_bps < 0) {
+    RTC_LOG(LS_WARNING) << "Config is invalid: min_bitrate_bps="
+                        << config_.min_bitrate_bps
+                        << "; max_bitrate_bps=" << config_.max_bitrate_bps
+                        << "; both expected greater or equal to 0";
+    return absl::nullopt;
+  }
+  TargetAudioBitrateConstraints constraints{
+      DataRate::BitsPerSec(config_.min_bitrate_bps),
+      DataRate::BitsPerSec(config_.max_bitrate_bps)};
+
+  // If bitrates were explicitly overriden via field trial, use those values.
+  if (allocation_settings_.min_bitrate)
+    constraints.min = *allocation_settings_.min_bitrate;
+  if (allocation_settings_.max_bitrate)
+    constraints.max = *allocation_settings_.max_bitrate;
+
+  RTC_DCHECK_GE(constraints.min, DataRate::Zero());
+  RTC_DCHECK_GE(constraints.max, DataRate::Zero());
+  if (constraints.max < constraints.min) {
+    RTC_LOG(LS_WARNING) << "TargetAudioBitrateConstraints::max is less than "
+                        << "TargetAudioBitrateConstraints::min";
+    return absl::nullopt;
+  }
+  if (use_legacy_overhead_calculation_) {
+    // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
+    const DataSize kOverheadPerPacket = DataSize::Bytes(20 + 8 + 10 + 12);
+    const TimeDelta kMaxFrameLength =
+        TimeDelta::Millis(60);  // Based on Opus spec
+    const DataRate kMinOverhead = kOverheadPerPacket / kMaxFrameLength;
+    constraints.min += kMinOverhead;
+    constraints.max += kMinOverhead;
+  } else {
+    if (!frame_length_range_.has_value()) {
+      RTC_LOG(LS_WARNING) << "frame_length_range_ is not set";
+      return absl::nullopt;
+    }
+    const DataSize kOverheadPerPacket =
+        DataSize::Bytes(total_packet_overhead_bytes_);
+    constraints.min += kOverheadPerPacket / frame_length_range_->second;
+    constraints.max += kOverheadPerPacket / frame_length_range_->first;
+  }
+  return constraints;
+}
+
+void AudioSendStream::RegisterCngPayloadType(int payload_type,
+                                             int clockrate_hz) {
+  channel_send_->RegisterCngPayloadType(payload_type, clockrate_hz);
+}
+
+void AudioSendStream::UpdateCachedTargetAudioBitrateConstraints() {
+  absl::optional<AudioSendStream::TargetAudioBitrateConstraints>
+      new_constraints = GetMinMaxBitrateConstraints();
+  if (!new_constraints.has_value()) {
+    return;
+  }
+  cached_constraints_ = new_constraints;
+}
+
+}  // namespace internal
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/audio_send_stream.h b/third_party/libwebrtc/audio/audio_send_stream.h
new file mode 100644
index 0000000000..62ccd524cb
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_send_stream.h
@@ -0,0 +1,238 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_AUDIO_SEND_STREAM_H_
+#define AUDIO_AUDIO_SEND_STREAM_H_
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "audio/audio_level.h"
+#include "audio/channel_send.h"
+#include "call/audio_send_stream.h"
+#include "call/audio_state.h"
+#include "call/bitrate_allocator.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+
+namespace webrtc {
+class RtcEventLog;
+class RtcpRttStats;
+class RtpTransportControllerSendInterface;
+
+struct AudioAllocationConfig {
+  static constexpr char kKey[] = "WebRTC-Audio-Allocation";
+  // Field Trial configured bitrates to use as overrides over default/user
+  // configured bitrate range when audio bitrate allocation is enabled.
+  absl::optional<DataRate> min_bitrate;
+  absl::optional<DataRate> max_bitrate;
+  DataRate priority_bitrate = DataRate::Zero();
+  // By default the priority_bitrate is compensated for packet overhead.
+  // Use this flag to configure a raw value instead.
+  absl::optional<DataRate> priority_bitrate_raw;
+  absl::optional<double> bitrate_priority;
+
+  std::unique_ptr<StructParametersParser> Parser();
+  explicit AudioAllocationConfig(const FieldTrialsView& field_trials);
+};
+namespace internal {
+class AudioState;
+
+class AudioSendStream final : public webrtc::AudioSendStream,
+                              public webrtc::BitrateAllocatorObserver {
+ public:
+  AudioSendStream(Clock* clock,
+                  const webrtc::AudioSendStream::Config& config,
+                  const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+                  TaskQueueFactory* task_queue_factory,
+                  RtpTransportControllerSendInterface* rtp_transport,
+                  BitrateAllocatorInterface* bitrate_allocator,
+                  RtcEventLog* event_log,
+                  RtcpRttStats* rtcp_rtt_stats,
+                  const absl::optional<RtpState>& suspended_rtp_state,
+                  const FieldTrialsView& field_trials);
+  // For unit tests, which need to supply a mock ChannelSend.
+  AudioSendStream(Clock* clock,
+                  const webrtc::AudioSendStream::Config& config,
+                  const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+                  TaskQueueFactory* task_queue_factory,
+                  RtpTransportControllerSendInterface* rtp_transport,
+                  BitrateAllocatorInterface* bitrate_allocator,
+                  RtcEventLog* event_log,
+                  const absl::optional<RtpState>& suspended_rtp_state,
+                  std::unique_ptr<voe::ChannelSendInterface> channel_send,
+                  const FieldTrialsView& field_trials);
+
+  AudioSendStream() = delete;
+  AudioSendStream(const AudioSendStream&) = delete;
+  AudioSendStream& operator=(const AudioSendStream&) = delete;
+
+  ~AudioSendStream() override;
+
+  // webrtc::AudioSendStream implementation.
+  const webrtc::AudioSendStream::Config& GetConfig() const override;
+  void Reconfigure(const webrtc::AudioSendStream::Config& config,
+                   SetParametersCallback callback) override;
+  void Start() override;
+  void Stop() override;
+  void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override;
+  bool SendTelephoneEvent(int payload_type,
+                          int payload_frequency,
+                          int event,
+                          int duration_ms) override;
+  void SetMuted(bool muted) override;
+  webrtc::AudioSendStream::Stats GetStats() const override;
+  webrtc::AudioSendStream::Stats GetStats(
+      bool has_remote_tracks) const override;
+
+  void DeliverRtcp(const uint8_t* packet, size_t length);
+
+  // Implements BitrateAllocatorObserver.
+  uint32_t OnBitrateUpdated(BitrateAllocationUpdate update) override;
+
+  void SetTransportOverhead(int transport_overhead_per_packet_bytes);
+
+  RtpState GetRtpState() const;
+  const voe::ChannelSendInterface* GetChannel() const;
+
+  // Returns combined per-packet overhead.
+  size_t TestOnlyGetPerPacketOverheadBytes() const
+      RTC_LOCKS_EXCLUDED(overhead_per_packet_lock_);
+
+ private:
+  class TimedTransport;
+  // Constraints including overhead.
+  struct TargetAudioBitrateConstraints {
+    DataRate min;
+    DataRate max;
+  };
+
+  internal::AudioState* audio_state();
+  const internal::AudioState* audio_state() const;
+
+  void StoreEncoderProperties(int sample_rate_hz, size_t num_channels)
+      RTC_RUN_ON(worker_thread_checker_);
+
+  void ConfigureStream(const Config& new_config,
+                       bool first_time,
+                       SetParametersCallback callback)
+      RTC_RUN_ON(worker_thread_checker_);
+  bool SetupSendCodec(const Config& new_config)
+      RTC_RUN_ON(worker_thread_checker_);
+  bool ReconfigureSendCodec(const Config& new_config)
+      RTC_RUN_ON(worker_thread_checker_);
+  void ReconfigureANA(const Config& new_config)
+      RTC_RUN_ON(worker_thread_checker_);
+  void ReconfigureCNG(const Config& new_config)
+      RTC_RUN_ON(worker_thread_checker_);
+  void ReconfigureBitrateObserver(const Config& new_config)
+      RTC_RUN_ON(worker_thread_checker_);
+
+  void ConfigureBitrateObserver() RTC_RUN_ON(worker_thread_checker_);
+  void RemoveBitrateObserver() RTC_RUN_ON(worker_thread_checker_);
+
+  // Returns bitrate constraints, maybe including overhead when enabled by
+  // field trial.
+  absl::optional<TargetAudioBitrateConstraints> GetMinMaxBitrateConstraints()
+      const RTC_RUN_ON(worker_thread_checker_);
+
+  // Sets per-packet overhead on encoded (for ANA) based on current known values
+  // of transport and packetization overheads.
+  void UpdateOverheadForEncoder()
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_);
+
+  // Returns combined per-packet overhead.
+  size_t GetPerPacketOverheadBytes() const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_);
+
+  void RegisterCngPayloadType(int payload_type, int clockrate_hz)
+      RTC_RUN_ON(worker_thread_checker_);
+
+  void UpdateCachedTargetAudioBitrateConstraints()
+      RTC_RUN_ON(worker_thread_checker_);
+
+  Clock* clock_;
+  const FieldTrialsView& field_trials_;
+
+  SequenceChecker worker_thread_checker_;
+  rtc::RaceChecker audio_capture_race_checker_;
+
+  const bool allocate_audio_without_feedback_;
+  const bool force_no_audio_feedback_ = allocate_audio_without_feedback_;
+  const bool enable_audio_alr_probing_;
+  const AudioAllocationConfig allocation_settings_;
+
+  webrtc::AudioSendStream::Config config_
+      RTC_GUARDED_BY(worker_thread_checker_);
+  rtc::scoped_refptr<webrtc::AudioState> audio_state_;
+  const std::unique_ptr<voe::ChannelSendInterface> channel_send_;
+  RtcEventLog* const event_log_;
+  const bool use_legacy_overhead_calculation_;
+
+  int encoder_sample_rate_hz_ RTC_GUARDED_BY(worker_thread_checker_) = 0;
+  size_t encoder_num_channels_ RTC_GUARDED_BY(worker_thread_checker_) = 0;
+  bool sending_ RTC_GUARDED_BY(worker_thread_checker_) = false;
+  mutable Mutex audio_level_lock_;
+  // Keeps track of audio level, total audio energy and total samples duration.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcaudiohandlerstats-totalaudioenergy
+  webrtc::voe::AudioLevel audio_level_ RTC_GUARDED_BY(audio_level_lock_);
+
+  BitrateAllocatorInterface* const bitrate_allocator_
+      RTC_GUARDED_BY(worker_thread_checker_);
+  absl::optional<AudioSendStream::TargetAudioBitrateConstraints>
+      cached_constraints_ RTC_GUARDED_BY(worker_thread_checker_) =
+          absl::nullopt;
+  RtpTransportControllerSendInterface* const rtp_transport_;
+
+  RtpRtcpInterface* const rtp_rtcp_module_;
+  absl::optional<RtpState> const suspended_rtp_state_;
+
+  // RFC 5285: Each distinct extension MUST have a unique ID. The value 0 is
+  // reserved for padding and MUST NOT be used as a local identifier.
+  // So it should be safe to use 0 here to indicate "not configured".
+  struct ExtensionIds {
+    int audio_level = 0;
+    int abs_send_time = 0;
+    int abs_capture_time = 0;
+    int transport_sequence_number = 0;
+    int mid = 0;
+    int rid = 0;
+    int repaired_rid = 0;
+  };
+  static ExtensionIds FindExtensionIds(
+      const std::vector<RtpExtension>& extensions);
+  static int TransportSeqNumId(const Config& config);
+
+  mutable Mutex overhead_per_packet_lock_;
+  size_t overhead_per_packet_ RTC_GUARDED_BY(overhead_per_packet_lock_) = 0;
+
+  // Current transport overhead (ICE, TURN, etc.)
+  size_t transport_overhead_per_packet_bytes_
+      RTC_GUARDED_BY(overhead_per_packet_lock_) = 0;
+
+  bool registered_with_allocator_ RTC_GUARDED_BY(worker_thread_checker_) =
+      false;
+  size_t total_packet_overhead_bytes_ RTC_GUARDED_BY(worker_thread_checker_) =
+      0;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> frame_length_range_
+      RTC_GUARDED_BY(worker_thread_checker_);
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_SEND_STREAM_H_
diff --git a/third_party/libwebrtc/audio/audio_send_stream_tests.cc b/third_party/libwebrtc/audio/audio_send_stream_tests.cc
new file mode 100644
index 0000000000..ff95ed70e1
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_send_stream_tests.cc
@@ -0,0 +1,249 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "test/call_test.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/rtcp_packet_parser.h"
+#include "test/video_test_constants.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+enum : int {  // The first valid value is 1.
+  kAudioLevelExtensionId = 1,
+  kTransportSequenceNumberExtensionId,
+};
+
+class AudioSendTest : public SendTest {
+ public:
+  AudioSendTest() : SendTest(VideoTestConstants::kDefaultTimeout) {}
+
+  size_t GetNumVideoStreams() const override { return 0; }
+  size_t GetNumAudioStreams() const override { return 1; }
+  size_t GetNumFlexfecStreams() const override { return 0; }
+};
+}  // namespace
+
+using AudioSendStreamCallTest = CallTest;
+
+TEST_F(AudioSendStreamCallTest, SupportsCName) {
+  static std::string kCName = "PjqatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
+  class CNameObserver : public AudioSendTest {
+   public:
+    CNameObserver() = default;
+
+   private:
+    Action OnSendRtcp(rtc::ArrayView<const uint8_t> packet) override {
+      RtcpPacketParser parser;
+      EXPECT_TRUE(parser.Parse(packet));
+      if (parser.sdes()->num_packets() > 0) {
+        EXPECT_EQ(1u, parser.sdes()->chunks().size());
+        EXPECT_EQ(kCName, parser.sdes()->chunks()[0].cname);
+
+        observation_complete_.Set();
+      }
+
+      return SEND_PACKET;
+    }
+
+    void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+                            std::vector<AudioReceiveStreamInterface::Config>*
+                                receive_configs) override {
+      send_config->rtp.c_name = kCName;
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for RTCP with CNAME.";
+    }
+  } test;
+
+  RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, NoExtensionsByDefault) {
+  class NoExtensionsObserver : public AudioSendTest {
+   public:
+    NoExtensionsObserver() = default;
+
+   private:
+    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
+      RtpPacket rtp_packet;
+      EXPECT_TRUE(rtp_packet.Parse(packet));          // rtp packet is valid.
+      EXPECT_EQ(packet[0] & 0b0001'0000, 0);          // extension bit not set.
+
+      observation_complete_.Set();
+      return SEND_PACKET;
+    }
+
+    void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+                            std::vector<AudioReceiveStreamInterface::Config>*
+                                receive_configs) override {
+      send_config->rtp.extensions.clear();
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
+    }
+  } test;
+
+  RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, SupportsAudioLevel) {
+  class AudioLevelObserver : public AudioSendTest {
+   public:
+    AudioLevelObserver() : AudioSendTest() {
+      extensions_.Register<AudioLevel>(kAudioLevelExtensionId);
+    }
+
+    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
+      RtpPacket rtp_packet(&extensions_);
+      EXPECT_TRUE(rtp_packet.Parse(packet));
+
+      uint8_t audio_level = 0;
+      bool voice = false;
+      EXPECT_TRUE(rtp_packet.GetExtension<AudioLevel>(&voice, &audio_level));
+      if (audio_level != 0) {
+        // Wait for at least one packet with a non-zero level.
+        observation_complete_.Set();
+      } else {
+        RTC_LOG(LS_WARNING) << "Got a packet with zero audioLevel - waiting"
+                               " for another packet...";
+      }
+
+      return SEND_PACKET;
+    }
+
+    void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+                            std::vector<AudioReceiveStreamInterface::Config>*
+                                receive_configs) override {
+      send_config->rtp.extensions.clear();
+      send_config->rtp.extensions.push_back(
+          RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelExtensionId));
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
+    }
+
+   private:
+    RtpHeaderExtensionMap extensions_;
+  } test;
+
+  RunBaseTest(&test);
+}
+
+class TransportWideSequenceNumberObserver : public AudioSendTest {
+ public:
+  explicit TransportWideSequenceNumberObserver(bool expect_sequence_number)
+      : AudioSendTest(), expect_sequence_number_(expect_sequence_number) {
+    extensions_.Register<TransportSequenceNumber>(
+        kTransportSequenceNumberExtensionId);
+  }
+
+ private:
+  Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
+    RtpPacket rtp_packet(&extensions_);
+    EXPECT_TRUE(rtp_packet.Parse(packet));
+
+    EXPECT_EQ(rtp_packet.HasExtension<TransportSequenceNumber>(),
+              expect_sequence_number_);
+    EXPECT_FALSE(rtp_packet.HasExtension<TransmissionOffset>());
+    EXPECT_FALSE(rtp_packet.HasExtension<AbsoluteSendTime>());
+
+    observation_complete_.Set();
+
+    return SEND_PACKET;
+  }
+
+  void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+                          std::vector<AudioReceiveStreamInterface::Config>*
+                              receive_configs) override {
+    send_config->rtp.extensions.clear();
+    send_config->rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                     kTransportSequenceNumberExtensionId));
+  }
+
+  void PerformTest() override {
+    EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
+  }
+  const bool expect_sequence_number_;
+  RtpHeaderExtensionMap extensions_;
+};
+
+TEST_F(AudioSendStreamCallTest, SendsTransportWideSequenceNumbersInFieldTrial) {
+  TransportWideSequenceNumberObserver test(/*expect_sequence_number=*/true);
+  RunBaseTest(&test);
+}
+
+TEST_F(AudioSendStreamCallTest, SendDtmf) {
+  static const uint8_t kDtmfPayloadType = 120;
+  static const int kDtmfPayloadFrequency = 8000;
+  static const int kDtmfEventFirst = 12;
+  static const int kDtmfEventLast = 31;
+  static const int kDtmfDuration = 50;
+  class DtmfObserver : public AudioSendTest {
+   public:
+    DtmfObserver() = default;
+
+   private:
+    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
+      RtpPacket rtp_packet;
+      EXPECT_TRUE(rtp_packet.Parse(packet));
+
+      if (rtp_packet.PayloadType() == kDtmfPayloadType) {
+        EXPECT_EQ(rtp_packet.headers_size(), 12u);
+        EXPECT_EQ(rtp_packet.size(), 16u);
+        const int event = rtp_packet.payload()[0];
+        if (event != expected_dtmf_event_) {
+          ++expected_dtmf_event_;
+          EXPECT_EQ(event, expected_dtmf_event_);
+          if (expected_dtmf_event_ == kDtmfEventLast) {
+            observation_complete_.Set();
+          }
+        }
+      }
+
+      return SEND_PACKET;
+    }
+
+    void OnAudioStreamsCreated(AudioSendStream* send_stream,
+                               const std::vector<AudioReceiveStreamInterface*>&
+                                   receive_streams) override {
+      // Need to start stream here, else DTMF events are dropped.
+      send_stream->Start();
+      for (int event = kDtmfEventFirst; event <= kDtmfEventLast; ++event) {
+        send_stream->SendTelephoneEvent(kDtmfPayloadType, kDtmfPayloadFrequency,
+                                        event, kDtmfDuration);
+      }
+    }
+
+    void PerformTest() override {
+      EXPECT_TRUE(Wait()) << "Timed out while waiting for DTMF stream.";
+    }
+
+    int expected_dtmf_event_ = kDtmfEventFirst;
+  } test;
+
+  RunBaseTest(&test);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/audio_send_stream_unittest.cc b/third_party/libwebrtc/audio/audio_send_stream_unittest.cc
new file mode 100644
index 0000000000..d842afdfe5
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_send_stream_unittest.cc
@@ -0,0 +1,927 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_send_stream.h"
+
+#include <memory>
+#include <string>
+#include <thread>
+#include <utility>
+#include <vector>
+
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/test/mock_frame_encryptor.h"
+#include "audio/audio_state.h"
+#include "audio/conversion.h"
+#include "audio/mock_voe_channel_proxy.h"
+#include "call/test/mock_rtp_transport_controller_send.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_mixer/sine_wave_generator.h"
+#include "modules/audio_processing/include/audio_processing_statistics.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "modules/rtp_rtcp/mocks/mock_network_link_rtcp_observer.h"
+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
+#include "test/mock_audio_encoder_factory.h"
+#include "test/scoped_key_value_config.h"
+#include "test/time_controller/real_time_controller.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using ::testing::_;
+using ::testing::AnyNumber;
+using ::testing::Eq;
+using ::testing::Field;
+using ::testing::InSequence;
+using ::testing::Invoke;
+using ::testing::Ne;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::StrEq;
+
+static const float kTolerance = 0.0001f;
+
+const uint32_t kSsrc = 1234;
+const char* kCName = "foo_name";
+const int kAudioLevelId = 2;
+const int kTransportSequenceNumberId = 4;
+const int32_t kEchoDelayMedian = 254;
+const int32_t kEchoDelayStdDev = -3;
+const double kDivergentFilterFraction = 0.2f;
+const double kEchoReturnLoss = -65;
+const double kEchoReturnLossEnhancement = 101;
+const double kResidualEchoLikelihood = -1.0f;
+const double kResidualEchoLikelihoodMax = 23.0f;
+const CallSendStatistics kCallStats = {112, 12, 13456, 17890};
+constexpr int kFractionLost = 123;
+constexpr int kCumulativeLost = 567;
+constexpr uint32_t kInterarrivalJitter = 132;
+const int kTelephoneEventPayloadType = 123;
+const int kTelephoneEventPayloadFrequency = 65432;
+const int kTelephoneEventCode = 45;
+const int kTelephoneEventDuration = 6789;
+constexpr int kIsacPayloadType = 103;
+const SdpAudioFormat kIsacFormat = {"isac", 16000, 1};
+const SdpAudioFormat kOpusFormat = {"opus", 48000, 2};
+const SdpAudioFormat kG722Format = {"g722", 8000, 1};
+const AudioCodecSpec kCodecSpecs[] = {
+    {kIsacFormat, {16000, 1, 32000, 10000, 32000}},
+    {kOpusFormat, {48000, 1, 32000, 6000, 510000}},
+    {kG722Format, {16000, 1, 64000}}};
+
+// TODO(dklee): This mirrors calculation in audio_send_stream.cc, which
+// should be made more precise in the future. This can be changed when that
+// logic is more accurate.
+const DataSize kOverheadPerPacket = DataSize::Bytes(20 + 8 + 10 + 12);
+const TimeDelta kMinFrameLength = TimeDelta::Millis(20);
+const TimeDelta kMaxFrameLength = TimeDelta::Millis(120);
+const DataRate kMinOverheadRate = kOverheadPerPacket / kMaxFrameLength;
+const DataRate kMaxOverheadRate = kOverheadPerPacket / kMinFrameLength;
+
+class MockLimitObserver : public BitrateAllocator::LimitObserver {
+ public:
+  MOCK_METHOD(void,
+              OnAllocationLimitsChanged,
+              (BitrateAllocationLimits),
+              (override));
+};
+
+std::unique_ptr<MockAudioEncoder> SetupAudioEncoderMock(
+    int payload_type,
+    const SdpAudioFormat& format) {
+  for (const auto& spec : kCodecSpecs) {
+    if (format == spec.format) {
+      std::unique_ptr<MockAudioEncoder> encoder(
+          new ::testing::NiceMock<MockAudioEncoder>());
+      ON_CALL(*encoder.get(), SampleRateHz())
+          .WillByDefault(Return(spec.info.sample_rate_hz));
+      ON_CALL(*encoder.get(), NumChannels())
+          .WillByDefault(Return(spec.info.num_channels));
+      ON_CALL(*encoder.get(), RtpTimestampRateHz())
+          .WillByDefault(Return(spec.format.clockrate_hz));
+      ON_CALL(*encoder.get(), GetFrameLengthRange())
+          .WillByDefault(Return(absl::optional<std::pair<TimeDelta, TimeDelta>>{
+              {TimeDelta::Millis(20), TimeDelta::Millis(120)}}));
+      return encoder;
+    }
+  }
+  return nullptr;
+}
+
+rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() {
+  rtc::scoped_refptr<MockAudioEncoderFactory> factory =
+      rtc::make_ref_counted<MockAudioEncoderFactory>();
+  ON_CALL(*factory.get(), GetSupportedEncoders())
+      .WillByDefault(Return(std::vector<AudioCodecSpec>(
+          std::begin(kCodecSpecs), std::end(kCodecSpecs))));
+  ON_CALL(*factory.get(), QueryAudioEncoder(_))
+      .WillByDefault(Invoke(
+          [](const SdpAudioFormat& format) -> absl::optional<AudioCodecInfo> {
+            for (const auto& spec : kCodecSpecs) {
+              if (format == spec.format) {
+                return spec.info;
+              }
+            }
+            return absl::nullopt;
+          }));
+  ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
+      .WillByDefault(Invoke([](int payload_type, const SdpAudioFormat& format,
+                               absl::optional<AudioCodecPairId> codec_pair_id,
+                               std::unique_ptr<AudioEncoder>* return_value) {
+        *return_value = SetupAudioEncoderMock(payload_type, format);
+      }));
+  return factory;
+}
+
+struct ConfigHelper {
+  ConfigHelper(bool audio_bwe_enabled,
+               bool expect_set_encoder_call,
+               bool use_null_audio_processing)
+      : stream_config_(/*send_transport=*/nullptr),
+        audio_processing_(
+            use_null_audio_processing
+                ? nullptr
+                : rtc::make_ref_counted<NiceMock<MockAudioProcessing>>()),
+        bitrate_allocator_(&limit_observer_),
+        audio_encoder_(nullptr) {
+    using ::testing::Invoke;
+
+    AudioState::Config config;
+    config.audio_mixer = AudioMixerImpl::Create();
+    config.audio_processing = audio_processing_;
+    config.audio_device_module = rtc::make_ref_counted<MockAudioDeviceModule>();
+    audio_state_ = AudioState::Create(config);
+
+    SetupDefaultChannelSend(audio_bwe_enabled);
+    SetupMockForSetupSendCodec(expect_set_encoder_call);
+    SetupMockForCallEncoder();
+
+    // Use ISAC as default codec so as to prevent unnecessary `channel_proxy_`
+    // calls from the default ctor behavior.
+    stream_config_.send_codec_spec =
+        AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
+    stream_config_.rtp.ssrc = kSsrc;
+    stream_config_.rtp.c_name = kCName;
+    stream_config_.rtp.extensions.push_back(
+        RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
+    if (audio_bwe_enabled) {
+      AddBweToConfig(&stream_config_);
+    }
+    stream_config_.encoder_factory = SetupEncoderFactoryMock();
+    stream_config_.min_bitrate_bps = 10000;
+    stream_config_.max_bitrate_bps = 65000;
+  }
+
+  std::unique_ptr<internal::AudioSendStream> CreateAudioSendStream() {
+    return std::unique_ptr<internal::AudioSendStream>(
+        new internal::AudioSendStream(
+            time_controller_.GetClock(), stream_config_, audio_state_,
+            time_controller_.GetTaskQueueFactory(), &rtp_transport_,
+            &bitrate_allocator_, &event_log_, absl::nullopt,
+            std::unique_ptr<voe::ChannelSendInterface>(channel_send_),
+            field_trials));
+  }
+
+  AudioSendStream::Config& config() { return stream_config_; }
+  MockAudioEncoderFactory& mock_encoder_factory() {
+    return *static_cast<MockAudioEncoderFactory*>(
+        stream_config_.encoder_factory.get());
+  }
+  MockRtpRtcpInterface* rtp_rtcp() { return &rtp_rtcp_; }
+  MockChannelSend* channel_send() { return channel_send_; }
+  RtpTransportControllerSendInterface* transport() { return &rtp_transport_; }
+
+  static void AddBweToConfig(AudioSendStream::Config* config) {
+    config->rtp.extensions.push_back(RtpExtension(
+        RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
+    config->send_codec_spec->transport_cc_enabled = true;
+  }
+
+  void SetupDefaultChannelSend(bool audio_bwe_enabled) {
+    EXPECT_TRUE(channel_send_ == nullptr);
+    channel_send_ = new ::testing::StrictMock<MockChannelSend>();
+    EXPECT_CALL(*channel_send_, GetRtpRtcp()).WillRepeatedly(Invoke([this]() {
+      return &this->rtp_rtcp_;
+    }));
+    EXPECT_CALL(rtp_rtcp_, SSRC).WillRepeatedly(Return(kSsrc));
+    EXPECT_CALL(*channel_send_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
+    EXPECT_CALL(*channel_send_, SetFrameEncryptor(_)).Times(1);
+    EXPECT_CALL(*channel_send_, SetEncoderToPacketizerFrameTransformer(_))
+        .Times(1);
+    EXPECT_CALL(rtp_rtcp_, SetExtmapAllowMixed(false)).Times(1);
+    EXPECT_CALL(*channel_send_,
+                SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
+        .Times(1);
+    EXPECT_CALL(rtp_transport_, GetRtcpObserver)
+        .WillRepeatedly(Return(&rtcp_observer_));
+    if (audio_bwe_enabled) {
+      EXPECT_CALL(rtp_rtcp_,
+                  RegisterRtpHeaderExtension(TransportSequenceNumber::Uri(),
+                                             kTransportSequenceNumberId))
+          .Times(1);
+    }
+    EXPECT_CALL(*channel_send_,
+                RegisterSenderCongestionControlObjects(&rtp_transport_))
+        .Times(1);
+    EXPECT_CALL(*channel_send_, ResetSenderCongestionControlObjects()).Times(1);
+  }
+
+  void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
+    if (expect_set_encoder_call) {
+      EXPECT_CALL(*channel_send_, SetEncoder)
+          .WillOnce(
+              [this](int payload_type, std::unique_ptr<AudioEncoder> encoder) {
+                this->audio_encoder_ = std::move(encoder);
+                return true;
+              });
+    }
+  }
+
+  void SetupMockForCallEncoder() {
+    // Let ModifyEncoder to invoke mock audio encoder.
+    EXPECT_CALL(*channel_send_, CallEncoder(_))
+        .WillRepeatedly(
+            [this](rtc::FunctionView<void(AudioEncoder*)> modifier) {
+              if (this->audio_encoder_)
+                modifier(this->audio_encoder_.get());
+            });
+  }
+
+  void SetupMockForSendTelephoneEvent() {
+    EXPECT_TRUE(channel_send_);
+    EXPECT_CALL(*channel_send_, SetSendTelephoneEventPayloadType(
+                                    kTelephoneEventPayloadType,
+                                    kTelephoneEventPayloadFrequency));
+    EXPECT_CALL(
+        *channel_send_,
+        SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
+        .WillOnce(Return(true));
+  }
+
+  void SetupMockForGetStats(bool use_null_audio_processing) {
+    using ::testing::DoAll;
+    using ::testing::SetArgPointee;
+    using ::testing::SetArgReferee;
+
+    std::vector<ReportBlockData> report_blocks;
+    ReportBlockData block;
+    block.set_source_ssrc(780);
+    block.set_fraction_lost_raw(kFractionLost);
+    block.set_cumulative_lost(kCumulativeLost);
+    block.set_jitter(kInterarrivalJitter);
+    report_blocks.push_back(block);  // Has wrong SSRC.
+    block.set_source_ssrc(kSsrc);
+    report_blocks.push_back(block);  // Correct block.
+    block.set_fraction_lost_raw(0);
+    report_blocks.push_back(block);  // Duplicate SSRC, bad fraction_lost.
+
+    EXPECT_TRUE(channel_send_);
+    EXPECT_CALL(*channel_send_, GetRTCPStatistics())
+        .WillRepeatedly(Return(kCallStats));
+    EXPECT_CALL(*channel_send_, GetRemoteRTCPReportBlocks())
+        .WillRepeatedly(Return(report_blocks));
+    EXPECT_CALL(*channel_send_, GetANAStatistics())
+        .WillRepeatedly(Return(ANAStats()));
+    EXPECT_CALL(*channel_send_, GetTargetBitrate()).WillRepeatedly(Return(0));
+
+    audio_processing_stats_.echo_return_loss = kEchoReturnLoss;
+    audio_processing_stats_.echo_return_loss_enhancement =
+        kEchoReturnLossEnhancement;
+    audio_processing_stats_.delay_median_ms = kEchoDelayMedian;
+    audio_processing_stats_.delay_standard_deviation_ms = kEchoDelayStdDev;
+    audio_processing_stats_.divergent_filter_fraction =
+        kDivergentFilterFraction;
+    audio_processing_stats_.residual_echo_likelihood = kResidualEchoLikelihood;
+    audio_processing_stats_.residual_echo_likelihood_recent_max =
+        kResidualEchoLikelihoodMax;
+    if (!use_null_audio_processing) {
+      ASSERT_TRUE(audio_processing_);
+      EXPECT_CALL(*audio_processing_, GetStatistics(true))
+          .WillRepeatedly(Return(audio_processing_stats_));
+    }
+  }
+
+  test::ScopedKeyValueConfig field_trials;
+
+ private:
+  RealTimeController time_controller_;
+  rtc::scoped_refptr<AudioState> audio_state_;
+  AudioSendStream::Config stream_config_;
+  ::testing::StrictMock<MockChannelSend>* channel_send_ = nullptr;
+  rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
+  AudioProcessingStats audio_processing_stats_;
+  ::testing::StrictMock<MockNetworkLinkRtcpObserver> rtcp_observer_;
+  ::testing::NiceMock<MockRtcEventLog> event_log_;
+  ::testing::NiceMock<MockRtpTransportControllerSend> rtp_transport_;
+  ::testing::NiceMock<MockRtpRtcpInterface> rtp_rtcp_;
+  ::testing::NiceMock<MockLimitObserver> limit_observer_;
+  BitrateAllocator bitrate_allocator_;
+  std::unique_ptr<AudioEncoder> audio_encoder_;
+};
+
+// The audio level ranges linearly [0,32767].
+std::unique_ptr<AudioFrame> CreateAudioFrame1kHzSineWave(int16_t audio_level,
+                                                         int duration_ms,
+                                                         int sample_rate_hz,
+                                                         size_t num_channels) {
+  size_t samples_per_channel = sample_rate_hz / (1000 / duration_ms);
+  std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
+  std::unique_ptr<AudioFrame> audio_frame = std::make_unique<AudioFrame>();
+  audio_frame->UpdateFrame(0 /* RTP timestamp */, &audio_data[0],
+                           samples_per_channel, sample_rate_hz,
+                           AudioFrame::SpeechType::kNormalSpeech,
+                           AudioFrame::VADActivity::kVadUnknown, num_channels);
+  SineWaveGenerator wave_generator(1000.0, audio_level);
+  wave_generator.GenerateNextFrame(audio_frame.get());
+  return audio_frame;
+}
+
+}  // namespace
+
+TEST(AudioSendStreamTest, ConfigToString) {
+  AudioSendStream::Config config(/*send_transport=*/nullptr);
+  config.rtp.ssrc = kSsrc;
+  config.rtp.c_name = kCName;
+  config.min_bitrate_bps = 12000;
+  config.max_bitrate_bps = 34000;
+  config.has_dscp = true;
+  config.send_codec_spec =
+      AudioSendStream::Config::SendCodecSpec(kIsacPayloadType, kIsacFormat);
+  config.send_codec_spec->nack_enabled = true;
+  config.send_codec_spec->transport_cc_enabled = false;
+  config.send_codec_spec->cng_payload_type = 42;
+  config.send_codec_spec->red_payload_type = 43;
+  config.encoder_factory = MockAudioEncoderFactory::CreateUnusedFactory();
+  config.rtp.extmap_allow_mixed = true;
+  config.rtp.extensions.push_back(
+      RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
+  config.rtcp_report_interval_ms = 2500;
+  EXPECT_EQ(
+      "{rtp: {ssrc: 1234, extmap-allow-mixed: true, extensions: [{uri: "
+      "urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 2}], "
+      "c_name: foo_name}, rtcp_report_interval_ms: 2500, "
+      "send_transport: null, "
+      "min_bitrate_bps: 12000, max_bitrate_bps: 34000, has "
+      "audio_network_adaptor_config: false, has_dscp: true, "
+      "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
+      "enable_non_sender_rtt: false, cng_payload_type: 42, "
+      "red_payload_type: 43, payload_type: 103, "
+      "format: {name: isac, clockrate_hz: 16000, num_channels: 1, "
+      "parameters: {}}}}",
+      config.ToString());
+}
+
+TEST(AudioSendStreamTest, ConstructDestruct) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+  }
+}
+
+TEST(AudioSendStreamTest, SendTelephoneEvent) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    helper.SetupMockForSendTelephoneEvent();
+    EXPECT_TRUE(send_stream->SendTelephoneEvent(
+        kTelephoneEventPayloadType, kTelephoneEventPayloadFrequency,
+        kTelephoneEventCode, kTelephoneEventDuration));
+  }
+}
+
+TEST(AudioSendStreamTest, SetMuted) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    EXPECT_CALL(*helper.channel_send(), SetInputMute(true));
+    send_stream->SetMuted(true);
+  }
+}
+
+TEST(AudioSendStreamTest, AudioBweCorrectObjectsOnChannelProxy) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+  }
+}
+
+TEST(AudioSendStreamTest, NoAudioBweCorrectObjectsOnChannelProxy) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+  }
+}
+
+TEST(AudioSendStreamTest, GetStats) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    helper.SetupMockForGetStats(use_null_audio_processing);
+    AudioSendStream::Stats stats = send_stream->GetStats(true);
+    EXPECT_EQ(kSsrc, stats.local_ssrc);
+    EXPECT_EQ(kCallStats.payload_bytes_sent, stats.payload_bytes_sent);
+    EXPECT_EQ(kCallStats.header_and_padding_bytes_sent,
+              stats.header_and_padding_bytes_sent);
+    EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
+    EXPECT_EQ(stats.packets_lost, kCumulativeLost);
+    EXPECT_FLOAT_EQ(stats.fraction_lost, Q8ToFloat(kFractionLost));
+    EXPECT_EQ(kIsacFormat.name, stats.codec_name);
+    EXPECT_EQ(stats.jitter_ms,
+              static_cast<int32_t>(kInterarrivalJitter /
+                                   (kIsacFormat.clockrate_hz / 1000)));
+    EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
+    EXPECT_EQ(0, stats.audio_level);
+    EXPECT_EQ(0, stats.total_input_energy);
+    EXPECT_EQ(0, stats.total_input_duration);
+
+    if (!use_null_audio_processing) {
+      EXPECT_EQ(kEchoDelayMedian, stats.apm_statistics.delay_median_ms);
+      EXPECT_EQ(kEchoDelayStdDev,
+                stats.apm_statistics.delay_standard_deviation_ms);
+      EXPECT_EQ(kEchoReturnLoss, stats.apm_statistics.echo_return_loss);
+      EXPECT_EQ(kEchoReturnLossEnhancement,
+                stats.apm_statistics.echo_return_loss_enhancement);
+      EXPECT_EQ(kDivergentFilterFraction,
+                stats.apm_statistics.divergent_filter_fraction);
+      EXPECT_EQ(kResidualEchoLikelihood,
+                stats.apm_statistics.residual_echo_likelihood);
+      EXPECT_EQ(kResidualEchoLikelihoodMax,
+                stats.apm_statistics.residual_echo_likelihood_recent_max);
+    }
+  }
+}
+
+TEST(AudioSendStreamTest, GetStatsAudioLevel) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    helper.SetupMockForGetStats(use_null_audio_processing);
+    EXPECT_CALL(*helper.channel_send(), ProcessAndEncodeAudio)
+        .Times(AnyNumber());
+
+    constexpr int kSampleRateHz = 48000;
+    constexpr size_t kNumChannels = 1;
+
+    constexpr int16_t kSilentAudioLevel = 0;
+    constexpr int16_t kMaxAudioLevel = 32767;  // Audio level is [0,32767].
+    constexpr int kAudioFrameDurationMs = 10;
+
+    // Process 10 audio frames (100 ms) of silence. After this, on the next
+    // (11-th) frame, the audio level will be updated with the maximum audio
+    // level of the first 11 frames. See AudioLevel.
+    for (size_t i = 0; i < 10; ++i) {
+      send_stream->SendAudioData(
+          CreateAudioFrame1kHzSineWave(kSilentAudioLevel, kAudioFrameDurationMs,
+                                       kSampleRateHz, kNumChannels));
+    }
+    AudioSendStream::Stats stats = send_stream->GetStats();
+    EXPECT_EQ(kSilentAudioLevel, stats.audio_level);
+    EXPECT_NEAR(0.0f, stats.total_input_energy, kTolerance);
+    EXPECT_NEAR(0.1f, stats.total_input_duration,
+                kTolerance);  // 100 ms = 0.1 s
+
+    // Process 10 audio frames (100 ms) of maximum audio level.
+    // Note that AudioLevel updates the audio level every 11th frame, processing
+    // 10 frames above was needed to see a non-zero audio level here.
+    for (size_t i = 0; i < 10; ++i) {
+      send_stream->SendAudioData(CreateAudioFrame1kHzSineWave(
+          kMaxAudioLevel, kAudioFrameDurationMs, kSampleRateHz, kNumChannels));
+    }
+    stats = send_stream->GetStats();
+    EXPECT_EQ(kMaxAudioLevel, stats.audio_level);
+    // Energy increases by energy*duration, where energy is audio level in
+    // [0,1].
+    EXPECT_NEAR(0.1f, stats.total_input_energy, kTolerance);  // 0.1 s of max
+    EXPECT_NEAR(0.2f, stats.total_input_duration,
+                kTolerance);  // 200 ms = 0.2 s
+  }
+}
+
+TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    helper.config().send_codec_spec =
+        AudioSendStream::Config::SendCodecSpec(0, kOpusFormat);
+    const std::string kAnaConfigString = "abcde";
+    const std::string kAnaReconfigString = "12345";
+
+    helper.config().audio_network_adaptor_config = kAnaConfigString;
+
+    EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _))
+        .WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString](
+                             int payload_type, const SdpAudioFormat& format,
+                             absl::optional<AudioCodecPairId> codec_pair_id,
+                             std::unique_ptr<AudioEncoder>* return_value) {
+          auto mock_encoder = SetupAudioEncoderMock(payload_type, format);
+          EXPECT_CALL(*mock_encoder,
+                      EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _))
+              .WillOnce(Return(true));
+          EXPECT_CALL(*mock_encoder,
+                      EnableAudioNetworkAdaptor(StrEq(kAnaReconfigString), _))
+              .WillOnce(Return(true));
+          *return_value = std::move(mock_encoder);
+        }));
+
+    auto send_stream = helper.CreateAudioSendStream();
+
+    auto stream_config = helper.config();
+    stream_config.audio_network_adaptor_config = kAnaReconfigString;
+
+    send_stream->Reconfigure(stream_config, nullptr);
+  }
+}
+
+TEST(AudioSendStreamTest, AudioNetworkAdaptorReceivesOverhead) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    helper.config().send_codec_spec =
+        AudioSendStream::Config::SendCodecSpec(0, kOpusFormat);
+    const std::string kAnaConfigString = "abcde";
+
+    EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _))
+        .WillOnce(Invoke(
+            [&kAnaConfigString](int payload_type, const SdpAudioFormat& format,
+                                absl::optional<AudioCodecPairId> codec_pair_id,
+                                std::unique_ptr<AudioEncoder>* return_value) {
+              auto mock_encoder = SetupAudioEncoderMock(payload_type, format);
+              InSequence s;
+              EXPECT_CALL(
+                  *mock_encoder,
+                  OnReceivedOverhead(Eq(kOverheadPerPacket.bytes<size_t>())))
+                  .Times(2);
+              EXPECT_CALL(*mock_encoder,
+                          EnableAudioNetworkAdaptor(StrEq(kAnaConfigString), _))
+                  .WillOnce(Return(true));
+              // Note: Overhead is received AFTER ANA has been enabled.
+              EXPECT_CALL(
+                  *mock_encoder,
+                  OnReceivedOverhead(Eq(kOverheadPerPacket.bytes<size_t>())))
+                  .WillOnce(Return());
+              *return_value = std::move(mock_encoder);
+            }));
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
+
+    auto send_stream = helper.CreateAudioSendStream();
+
+    auto stream_config = helper.config();
+    stream_config.audio_network_adaptor_config = kAnaConfigString;
+
+    send_stream->Reconfigure(stream_config, nullptr);
+  }
+}
+
+// VAD is applied when codec is mono and the CNG frequency matches the codec
+// clock rate.
+TEST(AudioSendStreamTest, SendCodecCanApplyVad) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, false, use_null_audio_processing);
+    helper.config().send_codec_spec =
+        AudioSendStream::Config::SendCodecSpec(9, kG722Format);
+    helper.config().send_codec_spec->cng_payload_type = 105;
+    std::unique_ptr<AudioEncoder> stolen_encoder;
+    EXPECT_CALL(*helper.channel_send(), SetEncoder)
+        .WillOnce([&stolen_encoder](int payload_type,
+                                    std::unique_ptr<AudioEncoder> encoder) {
+          stolen_encoder = std::move(encoder);
+          return true;
+        });
+    EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000));
+
+    auto send_stream = helper.CreateAudioSendStream();
+
+    // We cannot truly determine if the encoder created is an AudioEncoderCng.
+    // It is the only reasonable implementation that will return something from
+    // ReclaimContainedEncoders, though.
+    ASSERT_TRUE(stolen_encoder);
+    EXPECT_FALSE(stolen_encoder->ReclaimContainedEncoders().empty());
+  }
+}
+
+TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    EXPECT_CALL(
+        *helper.channel_send(),
+        OnBitrateAllocation(
+            Field(&BitrateAllocationUpdate::target_bitrate,
+                  Eq(DataRate::BitsPerSec(helper.config().max_bitrate_bps)))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps + 5000);
+    update.packet_loss_ratio = 0;
+    update.round_trip_time = TimeDelta::Millis(50);
+    update.bwe_period = TimeDelta::Millis(6000);
+    send_stream->OnBitrateUpdated(update);
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweTargetInRangeRespected) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    EXPECT_CALL(
+        *helper.channel_send(),
+        OnBitrateAllocation(Field(
+            &BitrateAllocationUpdate::target_bitrate,
+            Eq(DataRate::BitsPerSec(helper.config().max_bitrate_bps - 5000)))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps - 5000);
+    send_stream->OnBitrateUpdated(update);
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweFieldTrialMinRespected) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    ScopedKeyValueConfig field_trials(
+        helper.field_trials, "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
+    auto send_stream = helper.CreateAudioSendStream();
+    EXPECT_CALL(
+        *helper.channel_send(),
+        OnBitrateAllocation(Field(&BitrateAllocationUpdate::target_bitrate,
+                                  Eq(DataRate::KilobitsPerSec(6)))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate = DataRate::KilobitsPerSec(1);
+    send_stream->OnBitrateUpdated(update);
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweFieldTrialMaxRespected) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    ScopedKeyValueConfig field_trials(
+        helper.field_trials, "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
+    auto send_stream = helper.CreateAudioSendStream();
+    EXPECT_CALL(
+        *helper.channel_send(),
+        OnBitrateAllocation(Field(&BitrateAllocationUpdate::target_bitrate,
+                                  Eq(DataRate::KilobitsPerSec(64)))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate = DataRate::KilobitsPerSec(128);
+    send_stream->OnBitrateUpdated(update);
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweWithOverhead) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    ScopedKeyValueConfig field_trials(helper.field_trials,
+                                      "WebRTC-Audio-LegacyOverhead/Disabled/");
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
+    auto send_stream = helper.CreateAudioSendStream();
+    const DataRate bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps) +
+        kMaxOverheadRate;
+    EXPECT_CALL(*helper.channel_send(),
+                OnBitrateAllocation(Field(
+                    &BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate = bitrate;
+    send_stream->OnBitrateUpdated(update);
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweWithOverheadMinRespected) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    ScopedKeyValueConfig field_trials(
+        helper.field_trials,
+        "WebRTC-Audio-LegacyOverhead/Disabled/"
+        "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
+    auto send_stream = helper.CreateAudioSendStream();
+    const DataRate bitrate = DataRate::KilobitsPerSec(6) + kMinOverheadRate;
+    EXPECT_CALL(*helper.channel_send(),
+                OnBitrateAllocation(Field(
+                    &BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate = DataRate::KilobitsPerSec(1);
+    send_stream->OnBitrateUpdated(update);
+  }
+}
+
+TEST(AudioSendStreamTest, SSBweWithOverheadMaxRespected) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(true, true, use_null_audio_processing);
+    ScopedKeyValueConfig field_trials(
+        helper.field_trials,
+        "WebRTC-Audio-LegacyOverhead/Disabled/"
+        "WebRTC-Audio-Allocation/min:6kbps,max:64kbps/");
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(kOverheadPerPacket.bytes<size_t>()));
+    auto send_stream = helper.CreateAudioSendStream();
+    const DataRate bitrate = DataRate::KilobitsPerSec(64) + kMaxOverheadRate;
+    EXPECT_CALL(*helper.channel_send(),
+                OnBitrateAllocation(Field(
+                    &BitrateAllocationUpdate::target_bitrate, Eq(bitrate))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate = DataRate::KilobitsPerSec(128);
+    send_stream->OnBitrateUpdated(update);
+  }
+}
+
+TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+
+    EXPECT_CALL(*helper.channel_send(),
+                OnBitrateAllocation(Field(&BitrateAllocationUpdate::bwe_period,
+                                          Eq(TimeDelta::Millis(5000)))));
+    BitrateAllocationUpdate update;
+    update.target_bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps + 5000);
+    update.packet_loss_ratio = 0;
+    update.round_trip_time = TimeDelta::Millis(50);
+    update.bwe_period = TimeDelta::Millis(5000);
+    send_stream->OnBitrateUpdated(update);
+  }
+}
+
+// Test that AudioSendStream doesn't recreate the encoder unnecessarily.
+TEST(AudioSendStreamTest, DontRecreateEncoder) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, false, use_null_audio_processing);
+    // WillOnce is (currently) the default used by ConfigHelper if asked to set
+    // an expectation for SetEncoder. Since this behavior is essential for this
+    // test to be correct, it's instead set-up manually here. Otherwise a simple
+    // change to ConfigHelper (say to WillRepeatedly) would silently make this
+    // test useless.
+    EXPECT_CALL(*helper.channel_send(), SetEncoder).WillOnce(Return());
+
+    EXPECT_CALL(*helper.channel_send(), RegisterCngPayloadType(105, 8000));
+
+    helper.config().send_codec_spec =
+        AudioSendStream::Config::SendCodecSpec(9, kG722Format);
+    helper.config().send_codec_spec->cng_payload_type = 105;
+    auto send_stream = helper.CreateAudioSendStream();
+    send_stream->Reconfigure(helper.config(), nullptr);
+  }
+}
+
+TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+    ConfigHelper::AddBweToConfig(&new_config);
+
+    EXPECT_CALL(*helper.rtp_rtcp(),
+                RegisterRtpHeaderExtension(TransportSequenceNumber::Uri(),
+                                           kTransportSequenceNumberId))
+        .Times(1);
+    {
+      ::testing::InSequence seq;
+      EXPECT_CALL(*helper.channel_send(), ResetSenderCongestionControlObjects())
+          .Times(1);
+      EXPECT_CALL(*helper.channel_send(),
+                  RegisterSenderCongestionControlObjects(helper.transport()))
+          .Times(1);
+    }
+
+    send_stream->Reconfigure(new_config, nullptr);
+  }
+}
+
+TEST(AudioSendStreamTest, OnTransportOverheadChanged) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+
+    // CallEncoder will be called on overhead change.
+    EXPECT_CALL(*helper.channel_send(), CallEncoder);
+
+    const size_t transport_overhead_per_packet_bytes = 333;
+    send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
+
+    EXPECT_EQ(transport_overhead_per_packet_bytes,
+              send_stream->TestOnlyGetPerPacketOverheadBytes());
+  }
+}
+
+TEST(AudioSendStreamTest, DoesntCallEncoderWhenOverheadUnchanged) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+
+    // CallEncoder will be called on overhead change.
+    EXPECT_CALL(*helper.channel_send(), CallEncoder);
+    const size_t transport_overhead_per_packet_bytes = 333;
+    send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
+
+    // Set the same overhead again, CallEncoder should not be called again.
+    EXPECT_CALL(*helper.channel_send(), CallEncoder).Times(0);
+    send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
+
+    // New overhead, call CallEncoder again
+    EXPECT_CALL(*helper.channel_send(), CallEncoder);
+    send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes + 1);
+  }
+}
+
+TEST(AudioSendStreamTest, AudioOverheadChanged) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    const size_t audio_overhead_per_packet_bytes = 555;
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(audio_overhead_per_packet_bytes));
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+
+    BitrateAllocationUpdate update;
+    update.target_bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps) +
+        kMaxOverheadRate;
+    EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation);
+    send_stream->OnBitrateUpdated(update);
+
+    EXPECT_EQ(audio_overhead_per_packet_bytes,
+              send_stream->TestOnlyGetPerPacketOverheadBytes());
+
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(audio_overhead_per_packet_bytes + 20));
+    EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation);
+    send_stream->OnBitrateUpdated(update);
+
+    EXPECT_EQ(audio_overhead_per_packet_bytes + 20,
+              send_stream->TestOnlyGetPerPacketOverheadBytes());
+  }
+}
+
+TEST(AudioSendStreamTest, OnAudioAndTransportOverheadChanged) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    const size_t audio_overhead_per_packet_bytes = 555;
+    EXPECT_CALL(*helper.rtp_rtcp(), ExpectedPerPacketOverhead)
+        .WillRepeatedly(Return(audio_overhead_per_packet_bytes));
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+
+    const size_t transport_overhead_per_packet_bytes = 333;
+    send_stream->SetTransportOverhead(transport_overhead_per_packet_bytes);
+
+    BitrateAllocationUpdate update;
+    update.target_bitrate =
+        DataRate::BitsPerSec(helper.config().max_bitrate_bps) +
+        kMaxOverheadRate;
+    EXPECT_CALL(*helper.channel_send(), OnBitrateAllocation);
+    send_stream->OnBitrateUpdated(update);
+
+    EXPECT_EQ(
+        transport_overhead_per_packet_bytes + audio_overhead_per_packet_bytes,
+        send_stream->TestOnlyGetPerPacketOverheadBytes());
+  }
+}
+
+// Validates that reconfiguring the AudioSendStream with a Frame encryptor
+// correctly reconfigures on the object without crashing.
+TEST(AudioSendStreamTest, ReconfigureWithFrameEncryptor) {
+  for (bool use_null_audio_processing : {false, true}) {
+    ConfigHelper helper(false, true, use_null_audio_processing);
+    auto send_stream = helper.CreateAudioSendStream();
+    auto new_config = helper.config();
+
+    rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_0(
+        rtc::make_ref_counted<MockFrameEncryptor>());
+    new_config.frame_encryptor = mock_frame_encryptor_0;
+    EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr)))
+        .Times(1);
+    send_stream->Reconfigure(new_config, nullptr);
+
+    // Not updating the frame encryptor shouldn't force it to reconfigure.
+    EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(_)).Times(0);
+    send_stream->Reconfigure(new_config, nullptr);
+
+    // Updating frame encryptor to a new object should force a call to the
+    // proxy.
+    rtc::scoped_refptr<FrameEncryptorInterface> mock_frame_encryptor_1(
+        rtc::make_ref_counted<MockFrameEncryptor>());
+    new_config.frame_encryptor = mock_frame_encryptor_1;
+    new_config.crypto_options.sframe.require_frame_encryption = true;
+    EXPECT_CALL(*helper.channel_send(), SetFrameEncryptor(Ne(nullptr)))
+        .Times(1);
+    send_stream->Reconfigure(new_config, nullptr);
+  }
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/audio_state.cc b/third_party/libwebrtc/audio/audio_state.cc
new file mode 100644
index 0000000000..6f20e7b128
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_state.cc
@@ -0,0 +1,212 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_state.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/time_delta.h"
+#include "audio/audio_receive_stream.h"
+#include "audio/audio_send_stream.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace internal {
+
+AudioState::AudioState(const AudioState::Config& config)
+    : config_(config),
+      audio_transport_(config_.audio_mixer.get(),
+                       config_.audio_processing.get(),
+                       config_.async_audio_processing_factory.get()) {
+  RTC_DCHECK(config_.audio_mixer);
+  RTC_DCHECK(config_.audio_device_module);
+}
+
+AudioState::~AudioState() {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK(receiving_streams_.empty());
+  RTC_DCHECK(sending_streams_.empty());
+  RTC_DCHECK(!null_audio_poller_.Running());
+}
+
+AudioProcessing* AudioState::audio_processing() {
+  return config_.audio_processing.get();
+}
+
+AudioTransport* AudioState::audio_transport() {
+  return &audio_transport_;
+}
+
+void AudioState::AddReceivingStream(
+    webrtc::AudioReceiveStreamInterface* stream) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK_EQ(0, receiving_streams_.count(stream));
+  receiving_streams_.insert(stream);
+  if (!config_.audio_mixer->AddSource(
+          static_cast<AudioReceiveStreamImpl*>(stream))) {
+    RTC_DLOG(LS_ERROR) << "Failed to add source to mixer.";
+  }
+
+  // Make sure playback is initialized; start playing if enabled.
+  UpdateNullAudioPollerState();
+  auto* adm = config_.audio_device_module.get();
+  if (!adm->Playing()) {
+    if (adm->InitPlayout() == 0) {
+      if (playout_enabled_) {
+        adm->StartPlayout();
+      }
+    } else {
+      RTC_DLOG_F(LS_ERROR) << "Failed to initialize playout.";
+    }
+  }
+}
+
+void AudioState::RemoveReceivingStream(
+    webrtc::AudioReceiveStreamInterface* stream) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  auto count = receiving_streams_.erase(stream);
+  RTC_DCHECK_EQ(1, count);
+  config_.audio_mixer->RemoveSource(
+      static_cast<AudioReceiveStreamImpl*>(stream));
+  UpdateNullAudioPollerState();
+  if (receiving_streams_.empty()) {
+    config_.audio_device_module->StopPlayout();
+  }
+}
+
+void AudioState::AddSendingStream(webrtc::AudioSendStream* stream,
+                                  int sample_rate_hz,
+                                  size_t num_channels) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  auto& properties = sending_streams_[stream];
+  properties.sample_rate_hz = sample_rate_hz;
+  properties.num_channels = num_channels;
+  UpdateAudioTransportWithSendingStreams();
+
+  // Make sure recording is initialized; start recording if enabled.
+  auto* adm = config_.audio_device_module.get();
+  if (!adm->Recording()) {
+    if (adm->InitRecording() == 0) {
+      if (recording_enabled_) {
+        adm->StartRecording();
+      }
+    } else {
+      RTC_DLOG_F(LS_ERROR) << "Failed to initialize recording.";
+    }
+  }
+}
+
+void AudioState::RemoveSendingStream(webrtc::AudioSendStream* stream) {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  auto count = sending_streams_.erase(stream);
+  RTC_DCHECK_EQ(1, count);
+  UpdateAudioTransportWithSendingStreams();
+  if (sending_streams_.empty()) {
+    config_.audio_device_module->StopRecording();
+  }
+}
+
+void AudioState::SetPlayout(bool enabled) {
+  RTC_LOG(LS_INFO) << "SetPlayout(" << enabled << ")";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (playout_enabled_ != enabled) {
+    playout_enabled_ = enabled;
+    if (enabled) {
+      UpdateNullAudioPollerState();
+      if (!receiving_streams_.empty()) {
+        config_.audio_device_module->StartPlayout();
+      }
+    } else {
+      config_.audio_device_module->StopPlayout();
+      UpdateNullAudioPollerState();
+    }
+  }
+}
+
+void AudioState::SetRecording(bool enabled) {
+  RTC_LOG(LS_INFO) << "SetRecording(" << enabled << ")";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (recording_enabled_ != enabled) {
+    recording_enabled_ = enabled;
+    if (enabled) {
+      if (!sending_streams_.empty()) {
+        config_.audio_device_module->StartRecording();
+      }
+    } else {
+      config_.audio_device_module->StopRecording();
+    }
+  }
+}
+
+void AudioState::SetStereoChannelSwapping(bool enable) {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  audio_transport_.SetStereoChannelSwapping(enable);
+}
+
+void AudioState::UpdateAudioTransportWithSendingStreams() {
+  RTC_DCHECK(thread_checker_.IsCurrent());
+  std::vector<AudioSender*> audio_senders;
+  int max_sample_rate_hz = 8000;
+  size_t max_num_channels = 1;
+  for (const auto& kv : sending_streams_) {
+    audio_senders.push_back(kv.first);
+    max_sample_rate_hz = std::max(max_sample_rate_hz, kv.second.sample_rate_hz);
+    max_num_channels = std::max(max_num_channels, kv.second.num_channels);
+  }
+  audio_transport_.UpdateAudioSenders(std::move(audio_senders),
+                                      max_sample_rate_hz, max_num_channels);
+}
+
+void AudioState::UpdateNullAudioPollerState() {
+  // Run NullAudioPoller when there are receiving streams and playout is
+  // disabled.
+  if (!receiving_streams_.empty() && !playout_enabled_) {
+    if (!null_audio_poller_.Running()) {
+      AudioTransport* audio_transport = &audio_transport_;
+      null_audio_poller_ = RepeatingTaskHandle::Start(
+          TaskQueueBase::Current(), [audio_transport] {
+            static constexpr size_t kNumChannels = 1;
+            static constexpr uint32_t kSamplesPerSecond = 48'000;
+            // 10ms of samples
+            static constexpr size_t kNumSamples = kSamplesPerSecond / 100;
+
+            // Buffer to hold the audio samples.
+            int16_t buffer[kNumSamples * kNumChannels];
+
+            // Output variables from `NeedMorePlayData`.
+            size_t n_samples;
+            int64_t elapsed_time_ms;
+            int64_t ntp_time_ms;
+            audio_transport->NeedMorePlayData(
+                kNumSamples, sizeof(int16_t), kNumChannels, kSamplesPerSecond,
+                buffer, n_samples, &elapsed_time_ms, &ntp_time_ms);
+
+            // Reschedule the next poll iteration.
+            return TimeDelta::Millis(10);
+          });
+    }
+  } else {
+    null_audio_poller_.Stop();
+  }
+}
+}  // namespace internal
+
+rtc::scoped_refptr<AudioState> AudioState::Create(
+    const AudioState::Config& config) {
+  return rtc::make_ref_counted<internal::AudioState>(config);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/audio_state.h b/third_party/libwebrtc/audio/audio_state.h
new file mode 100644
index 0000000000..88aaaa3697
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_state.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_AUDIO_STATE_H_
+#define AUDIO_AUDIO_STATE_H_
+
+#include <map>
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "audio/audio_transport_impl.h"
+#include "call/audio_state.h"
+#include "rtc_base/containers/flat_set.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioSendStream;
+class AudioReceiveStreamInterface;
+
+namespace internal {
+
+class AudioState : public webrtc::AudioState {
+ public:
+  explicit AudioState(const AudioState::Config& config);
+
+  AudioState() = delete;
+  AudioState(const AudioState&) = delete;
+  AudioState& operator=(const AudioState&) = delete;
+
+  ~AudioState() override;
+
+  AudioProcessing* audio_processing() override;
+  AudioTransport* audio_transport() override;
+
+  void SetPlayout(bool enabled) override;
+  void SetRecording(bool enabled) override;
+
+  void SetStereoChannelSwapping(bool enable) override;
+
+  AudioDeviceModule* audio_device_module() {
+    RTC_DCHECK(config_.audio_device_module);
+    return config_.audio_device_module.get();
+  }
+
+  void AddReceivingStream(webrtc::AudioReceiveStreamInterface* stream);
+  void RemoveReceivingStream(webrtc::AudioReceiveStreamInterface* stream);
+
+  void AddSendingStream(webrtc::AudioSendStream* stream,
+                        int sample_rate_hz,
+                        size_t num_channels);
+  void RemoveSendingStream(webrtc::AudioSendStream* stream);
+
+ private:
+  void UpdateAudioTransportWithSendingStreams();
+  void UpdateNullAudioPollerState() RTC_RUN_ON(&thread_checker_);
+
+  SequenceChecker thread_checker_;
+  SequenceChecker process_thread_checker_{SequenceChecker::kDetached};
+  const webrtc::AudioState::Config config_;
+  bool recording_enabled_ = true;
+  bool playout_enabled_ = true;
+
+  // Transports mixed audio from the mixer to the audio device and
+  // recorded audio to the sending streams.
+  AudioTransportImpl audio_transport_;
+
+  // Null audio poller is used to continue polling the audio streams if audio
+  // playout is disabled so that audio processing still happens and the audio
+  // stats are still updated.
+  RepeatingTaskHandle null_audio_poller_ RTC_GUARDED_BY(&thread_checker_);
+
+  webrtc::flat_set<webrtc::AudioReceiveStreamInterface*> receiving_streams_;
+  struct StreamProperties {
+    int sample_rate_hz = 0;
+    size_t num_channels = 0;
+  };
+  std::map<webrtc::AudioSendStream*, StreamProperties> sending_streams_;
+};
+}  // namespace internal
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_STATE_H_
diff --git a/third_party/libwebrtc/audio/audio_state_unittest.cc b/third_party/libwebrtc/audio/audio_state_unittest.cc
new file mode 100644
index 0000000000..070e220979
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_state_unittest.cc
@@ -0,0 +1,368 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_state.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/task_queue/test/mock_task_queue_base.h"
+#include "call/test/mock_audio_send_stream.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using ::testing::_;
+using ::testing::Matcher;
+using ::testing::NiceMock;
+using ::testing::StrictMock;
+using ::testing::Values;
+
+constexpr int kSampleRate = 16000;
+constexpr int kNumberOfChannels = 1;
+
+struct FakeAsyncAudioProcessingHelper {
+  class FakeTaskQueue : public StrictMock<MockTaskQueueBase> {
+   public:
+    FakeTaskQueue() = default;
+
+    void Delete() override { delete this; }
+    void PostTaskImpl(absl::AnyInvocable<void() &&> task,
+                      const PostTaskTraits& /*traits*/,
+                      const Location& /*location*/) override {
+      std::move(task)();
+    }
+  };
+
+  class FakeTaskQueueFactory : public TaskQueueFactory {
+   public:
+    FakeTaskQueueFactory() = default;
+    ~FakeTaskQueueFactory() override = default;
+    std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
+        absl::string_view name,
+        Priority priority) const override {
+      return std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter>(
+          new FakeTaskQueue());
+    }
+  };
+
+  class MockAudioFrameProcessor : public AudioFrameProcessor {
+   public:
+    ~MockAudioFrameProcessor() override = default;
+
+    MOCK_METHOD(void, ProcessCalled, ());
+    MOCK_METHOD(void, SinkSet, ());
+    MOCK_METHOD(void, SinkCleared, ());
+
+    void Process(std::unique_ptr<AudioFrame> frame) override {
+      ProcessCalled();
+      sink_callback_(std::move(frame));
+    }
+
+    void SetSink(OnAudioFrameCallback sink_callback) override {
+      sink_callback_ = std::move(sink_callback);
+      if (sink_callback_ == nullptr)
+        SinkCleared();
+      else
+        SinkSet();
+    }
+
+   private:
+    OnAudioFrameCallback sink_callback_;
+  };
+
+  NiceMock<MockAudioFrameProcessor> audio_frame_processor_;
+  FakeTaskQueueFactory task_queue_factory_;
+
+  rtc::scoped_refptr<AsyncAudioProcessing::Factory> CreateFactory() {
+    return rtc::make_ref_counted<AsyncAudioProcessing::Factory>(
+        audio_frame_processor_, task_queue_factory_);
+  }
+};
+
+struct ConfigHelper {
+  struct Params {
+    bool use_null_audio_processing;
+    bool use_async_audio_processing;
+  };
+
+  explicit ConfigHelper(const Params& params)
+      : audio_mixer(AudioMixerImpl::Create()) {
+    audio_state_config.audio_mixer = audio_mixer;
+    audio_state_config.audio_processing =
+        params.use_null_audio_processing
+            ? nullptr
+            : rtc::make_ref_counted<testing::NiceMock<MockAudioProcessing>>();
+    audio_state_config.audio_device_module =
+        rtc::make_ref_counted<NiceMock<MockAudioDeviceModule>>();
+    if (params.use_async_audio_processing) {
+      audio_state_config.async_audio_processing_factory =
+          async_audio_processing_helper_.CreateFactory();
+    }
+  }
+  AudioState::Config& config() { return audio_state_config; }
+  rtc::scoped_refptr<AudioMixer> mixer() { return audio_mixer; }
+  NiceMock<FakeAsyncAudioProcessingHelper::MockAudioFrameProcessor>&
+  mock_audio_frame_processor() {
+    return async_audio_processing_helper_.audio_frame_processor_;
+  }
+
+ private:
+  AudioState::Config audio_state_config;
+  rtc::scoped_refptr<AudioMixer> audio_mixer;
+  FakeAsyncAudioProcessingHelper async_audio_processing_helper_;
+};
+
+class FakeAudioSource : public AudioMixer::Source {
+ public:
+  // TODO(aleloi): Valid overrides commented out, because the gmock
+  // methods don't use any override declarations, and we want to avoid
+  // warnings from -Winconsistent-missing-override. See
+  // http://crbug.com/428099.
+  int Ssrc() const /*override*/ { return 0; }
+
+  int PreferredSampleRate() const /*override*/ { return kSampleRate; }
+
+  MOCK_METHOD(AudioFrameInfo,
+              GetAudioFrameWithInfo,
+              (int sample_rate_hz, AudioFrame*),
+              (override));
+};
+
+std::vector<int16_t> Create10msTestData(int sample_rate_hz,
+                                        size_t num_channels) {
+  const int samples_per_channel = sample_rate_hz / 100;
+  std::vector<int16_t> audio_data(samples_per_channel * num_channels, 0);
+  // Fill the first channel with a 1kHz sine wave.
+  const float inc = (2 * 3.14159265f * 1000) / sample_rate_hz;
+  float w = 0.f;
+  for (int i = 0; i < samples_per_channel; ++i) {
+    audio_data[i * num_channels] = static_cast<int16_t>(32767.f * std::sin(w));
+    w += inc;
+  }
+  return audio_data;
+}
+
+std::vector<uint32_t> ComputeChannelLevels(AudioFrame* audio_frame) {
+  const size_t num_channels = audio_frame->num_channels_;
+  const size_t samples_per_channel = audio_frame->samples_per_channel_;
+  std::vector<uint32_t> levels(num_channels, 0);
+  for (size_t i = 0; i < samples_per_channel; ++i) {
+    for (size_t j = 0; j < num_channels; ++j) {
+      levels[j] += std::abs(audio_frame->data()[i * num_channels + j]);
+    }
+  }
+  return levels;
+}
+}  // namespace
+
+class AudioStateTest : public ::testing::TestWithParam<ConfigHelper::Params> {};
+
+TEST_P(AudioStateTest, Create) {
+  ConfigHelper helper(GetParam());
+  auto audio_state = AudioState::Create(helper.config());
+  EXPECT_TRUE(audio_state.get());
+}
+
+TEST_P(AudioStateTest, ConstructDestruct) {
+  ConfigHelper helper(GetParam());
+  rtc::scoped_refptr<internal::AudioState> audio_state(
+      rtc::make_ref_counted<internal::AudioState>(helper.config()));
+}
+
+TEST_P(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
+  ConfigHelper helper(GetParam());
+
+  if (GetParam().use_async_audio_processing) {
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
+  }
+
+  rtc::scoped_refptr<internal::AudioState> audio_state(
+      rtc::make_ref_counted<internal::AudioState>(helper.config()));
+
+  MockAudioSendStream stream;
+  audio_state->AddSendingStream(&stream, 8000, 2);
+
+  EXPECT_CALL(
+      stream,
+      SendAudioDataForMock(::testing::AllOf(
+          ::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(8000)),
+          ::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(2u)))))
+      .WillOnce(
+          // Verify that channels are not swapped by default.
+          ::testing::Invoke([](AudioFrame* audio_frame) {
+            auto levels = ComputeChannelLevels(audio_frame);
+            EXPECT_LT(0u, levels[0]);
+            EXPECT_EQ(0u, levels[1]);
+          }));
+  MockAudioProcessing* ap =
+      GetParam().use_null_audio_processing
+          ? nullptr
+          : static_cast<MockAudioProcessing*>(audio_state->audio_processing());
+  if (ap) {
+    EXPECT_CALL(*ap, set_stream_delay_ms(0));
+    EXPECT_CALL(*ap, set_stream_key_pressed(false));
+    EXPECT_CALL(*ap, ProcessStream(_, _, _, Matcher<int16_t*>(_)));
+  }
+
+  constexpr int kSampleRate = 16000;
+  constexpr size_t kNumChannels = 2;
+  auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
+  uint32_t new_mic_level = 667;
+  audio_state->audio_transport()->RecordedDataIsAvailable(
+      &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
+      kSampleRate, 0, 0, 0, false, new_mic_level);
+  EXPECT_EQ(667u, new_mic_level);
+
+  audio_state->RemoveSendingStream(&stream);
+}
+
+TEST_P(AudioStateTest, RecordedAudioArrivesAtMultipleStreams) {
+  ConfigHelper helper(GetParam());
+
+  if (GetParam().use_async_audio_processing) {
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
+  }
+
+  rtc::scoped_refptr<internal::AudioState> audio_state(
+      rtc::make_ref_counted<internal::AudioState>(helper.config()));
+
+  MockAudioSendStream stream_1;
+  MockAudioSendStream stream_2;
+  audio_state->AddSendingStream(&stream_1, 8001, 2);
+  audio_state->AddSendingStream(&stream_2, 32000, 1);
+
+  EXPECT_CALL(
+      stream_1,
+      SendAudioDataForMock(::testing::AllOf(
+          ::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(16000)),
+          ::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(1u)))))
+      .WillOnce(
+          // Verify that there is output signal.
+          ::testing::Invoke([](AudioFrame* audio_frame) {
+            auto levels = ComputeChannelLevels(audio_frame);
+            EXPECT_LT(0u, levels[0]);
+          }));
+  EXPECT_CALL(
+      stream_2,
+      SendAudioDataForMock(::testing::AllOf(
+          ::testing::Field(&AudioFrame::sample_rate_hz_, ::testing::Eq(16000)),
+          ::testing::Field(&AudioFrame::num_channels_, ::testing::Eq(1u)))))
+      .WillOnce(
+          // Verify that there is output signal.
+          ::testing::Invoke([](AudioFrame* audio_frame) {
+            auto levels = ComputeChannelLevels(audio_frame);
+            EXPECT_LT(0u, levels[0]);
+          }));
+  MockAudioProcessing* ap =
+      static_cast<MockAudioProcessing*>(audio_state->audio_processing());
+  if (ap) {
+    EXPECT_CALL(*ap, set_stream_delay_ms(5));
+    EXPECT_CALL(*ap, set_stream_key_pressed(true));
+    EXPECT_CALL(*ap, ProcessStream(_, _, _, Matcher<int16_t*>(_)));
+  }
+
+  constexpr int kSampleRate = 16000;
+  constexpr size_t kNumChannels = 1;
+  auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
+  uint32_t new_mic_level = 667;
+  audio_state->audio_transport()->RecordedDataIsAvailable(
+      &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
+      kSampleRate, 5, 0, 0, true, new_mic_level);
+  EXPECT_EQ(667u, new_mic_level);
+
+  audio_state->RemoveSendingStream(&stream_1);
+  audio_state->RemoveSendingStream(&stream_2);
+}
+
+TEST_P(AudioStateTest, EnableChannelSwap) {
+  constexpr int kSampleRate = 16000;
+  constexpr size_t kNumChannels = 2;
+
+  ConfigHelper helper(GetParam());
+
+  if (GetParam().use_async_audio_processing) {
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkSet);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), ProcessCalled);
+    EXPECT_CALL(helper.mock_audio_frame_processor(), SinkCleared);
+  }
+
+  rtc::scoped_refptr<internal::AudioState> audio_state(
+      rtc::make_ref_counted<internal::AudioState>(helper.config()));
+
+  audio_state->SetStereoChannelSwapping(true);
+
+  MockAudioSendStream stream;
+  audio_state->AddSendingStream(&stream, kSampleRate, kNumChannels);
+
+  EXPECT_CALL(stream, SendAudioDataForMock(_))
+      .WillOnce(
+          // Verify that channels are swapped.
+          ::testing::Invoke([](AudioFrame* audio_frame) {
+            auto levels = ComputeChannelLevels(audio_frame);
+            EXPECT_EQ(0u, levels[0]);
+            EXPECT_LT(0u, levels[1]);
+          }));
+
+  auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
+  uint32_t new_mic_level = 667;
+  audio_state->audio_transport()->RecordedDataIsAvailable(
+      &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
+      kSampleRate, 0, 0, 0, false, new_mic_level);
+  EXPECT_EQ(667u, new_mic_level);
+
+  audio_state->RemoveSendingStream(&stream);
+}
+
+TEST_P(AudioStateTest,
+       QueryingTransportForAudioShouldResultInGetAudioCallOnMixerSource) {
+  ConfigHelper helper(GetParam());
+  auto audio_state = AudioState::Create(helper.config());
+
+  FakeAudioSource fake_source;
+  helper.mixer()->AddSource(&fake_source);
+
+  EXPECT_CALL(fake_source, GetAudioFrameWithInfo(_, _))
+      .WillOnce(
+          ::testing::Invoke([](int sample_rate_hz, AudioFrame* audio_frame) {
+            audio_frame->sample_rate_hz_ = sample_rate_hz;
+            audio_frame->samples_per_channel_ = sample_rate_hz / 100;
+            audio_frame->num_channels_ = kNumberOfChannels;
+            return AudioMixer::Source::AudioFrameInfo::kNormal;
+          }));
+
+  int16_t audio_buffer[kSampleRate / 100 * kNumberOfChannels];
+  size_t n_samples_out;
+  int64_t elapsed_time_ms;
+  int64_t ntp_time_ms;
+  audio_state->audio_transport()->NeedMorePlayData(
+      kSampleRate / 100, kNumberOfChannels * 2, kNumberOfChannels, kSampleRate,
+      audio_buffer, n_samples_out, &elapsed_time_ms, &ntp_time_ms);
+}
+
+INSTANTIATE_TEST_SUITE_P(AudioStateTest,
+                         AudioStateTest,
+                         Values(ConfigHelper::Params({false, false}),
+                                ConfigHelper::Params({true, false}),
+                                ConfigHelper::Params({false, true}),
+                                ConfigHelper::Params({true, true})));
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/audio_transport_impl.cc b/third_party/libwebrtc/audio/audio_transport_impl.cc
new file mode 100644
index 0000000000..42a81d5b4a
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_transport_impl.cc
@@ -0,0 +1,285 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/audio_transport_impl.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "audio/remix_resample.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "call/audio_sender.h"
+#include "modules/async_audio_processing/async_audio_processing.h"
+#include "modules/audio_processing/include/audio_frame_proxies.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+namespace {
+
+// We want to process at the lowest sample rate and channel count possible
+// without losing information. Choose the lowest native rate at least equal to
+// the minimum of input and codec rates, choose lowest channel count, and
+// configure the audio frame.
+void InitializeCaptureFrame(int input_sample_rate,
+                            int send_sample_rate_hz,
+                            size_t input_num_channels,
+                            size_t send_num_channels,
+                            AudioFrame* audio_frame) {
+  RTC_DCHECK(audio_frame);
+  int min_processing_rate_hz = std::min(input_sample_rate, send_sample_rate_hz);
+  for (int native_rate_hz : AudioProcessing::kNativeSampleRatesHz) {
+    audio_frame->sample_rate_hz_ = native_rate_hz;
+    if (audio_frame->sample_rate_hz_ >= min_processing_rate_hz) {
+      break;
+    }
+  }
+  audio_frame->num_channels_ = std::min(input_num_channels, send_num_channels);
+}
+
+void ProcessCaptureFrame(uint32_t delay_ms,
+                         bool key_pressed,
+                         bool swap_stereo_channels,
+                         AudioProcessing* audio_processing,
+                         AudioFrame* audio_frame) {
+  RTC_DCHECK(audio_frame);
+  if (audio_processing) {
+    audio_processing->set_stream_delay_ms(delay_ms);
+    audio_processing->set_stream_key_pressed(key_pressed);
+    int error = ProcessAudioFrame(audio_processing, audio_frame);
+
+    RTC_DCHECK_EQ(0, error) << "ProcessStream() error: " << error;
+  }
+
+  if (swap_stereo_channels) {
+    AudioFrameOperations::SwapStereoChannels(audio_frame);
+  }
+}
+
+// Resample audio in `frame` to given sample rate preserving the
+// channel count and place the result in `destination`.
+int Resample(const AudioFrame& frame,
+             const int destination_sample_rate,
+             PushResampler<int16_t>* resampler,
+             int16_t* destination) {
+  TRACE_EVENT2("webrtc", "Resample", "frame sample rate", frame.sample_rate_hz_,
+               "destination_sample_rate", destination_sample_rate);
+  const int number_of_channels = static_cast<int>(frame.num_channels_);
+  const int target_number_of_samples_per_channel =
+      destination_sample_rate / 100;
+  resampler->InitializeIfNeeded(frame.sample_rate_hz_, destination_sample_rate,
+                                number_of_channels);
+
+  // TODO(yujo): make resampler take an AudioFrame, and add special case
+  // handling of muted frames.
+  return resampler->Resample(
+      frame.data(), frame.samples_per_channel_ * number_of_channels,
+      destination, number_of_channels * target_number_of_samples_per_channel);
+}
+}  // namespace
+
+AudioTransportImpl::AudioTransportImpl(
+    AudioMixer* mixer,
+    AudioProcessing* audio_processing,
+    AsyncAudioProcessing::Factory* async_audio_processing_factory)
+    : audio_processing_(audio_processing),
+      async_audio_processing_(
+          async_audio_processing_factory
+              ? async_audio_processing_factory->CreateAsyncAudioProcessing(
+                    [this](std::unique_ptr<AudioFrame> frame) {
+                      this->SendProcessedData(std::move(frame));
+                    })
+              : nullptr),
+      mixer_(mixer) {
+  RTC_DCHECK(mixer);
+}
+
+AudioTransportImpl::~AudioTransportImpl() {}
+
+int32_t AudioTransportImpl::RecordedDataIsAvailable(
+    const void* audio_data,
+    size_t number_of_frames,
+    size_t bytes_per_sample,
+    size_t number_of_channels,
+    uint32_t sample_rate,
+    uint32_t audio_delay_milliseconds,
+    int32_t clock_drift,
+    uint32_t volume,
+    bool key_pressed,
+    uint32_t& new_mic_volume) {  // NOLINT: to avoid changing APIs
+  return RecordedDataIsAvailable(
+      audio_data, number_of_frames, bytes_per_sample, number_of_channels,
+      sample_rate, audio_delay_milliseconds, clock_drift, volume, key_pressed,
+      new_mic_volume, /*estimated_capture_time_ns=*/absl::nullopt);
+}
+
+// Not used in Chromium. Process captured audio and distribute to all sending
+// streams, and try to do this at the lowest possible sample rate.
+int32_t AudioTransportImpl::RecordedDataIsAvailable(
+    const void* audio_data,
+    size_t number_of_frames,
+    size_t bytes_per_sample,
+    size_t number_of_channels,
+    uint32_t sample_rate,
+    uint32_t audio_delay_milliseconds,
+    int32_t /*clock_drift*/,
+    uint32_t /*volume*/,
+    bool key_pressed,
+    uint32_t& /*new_mic_volume*/,
+    absl::optional<int64_t>
+        estimated_capture_time_ns) {  // NOLINT: to avoid changing APIs
+  RTC_DCHECK(audio_data);
+  RTC_DCHECK_GE(number_of_channels, 1);
+  RTC_DCHECK_LE(number_of_channels, 2);
+  RTC_DCHECK_EQ(2 * number_of_channels, bytes_per_sample);
+  RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
+  // 100 = 1 second / data duration (10 ms).
+  RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
+  RTC_DCHECK_LE(bytes_per_sample * number_of_frames * number_of_channels,
+                AudioFrame::kMaxDataSizeBytes);
+
+  int send_sample_rate_hz = 0;
+  size_t send_num_channels = 0;
+  bool swap_stereo_channels = false;
+  {
+    MutexLock lock(&capture_lock_);
+    send_sample_rate_hz = send_sample_rate_hz_;
+    send_num_channels = send_num_channels_;
+    swap_stereo_channels = swap_stereo_channels_;
+  }
+
+  std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
+  InitializeCaptureFrame(sample_rate, send_sample_rate_hz, number_of_channels,
+                         send_num_channels, audio_frame.get());
+  voe::RemixAndResample(static_cast<const int16_t*>(audio_data),
+                        number_of_frames, number_of_channels, sample_rate,
+                        &capture_resampler_, audio_frame.get());
+  ProcessCaptureFrame(audio_delay_milliseconds, key_pressed,
+                      swap_stereo_channels, audio_processing_,
+                      audio_frame.get());
+
+  if (estimated_capture_time_ns) {
+    audio_frame->set_absolute_capture_timestamp_ms(*estimated_capture_time_ns /
+                                                   1000000);
+  }
+
+  RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
+  if (async_audio_processing_)
+    async_audio_processing_->Process(std::move(audio_frame));
+  else
+    SendProcessedData(std::move(audio_frame));
+
+  return 0;
+}
+
+void AudioTransportImpl::SendProcessedData(
+    std::unique_ptr<AudioFrame> audio_frame) {
+  TRACE_EVENT0("webrtc", "AudioTransportImpl::SendProcessedData");
+  RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
+  MutexLock lock(&capture_lock_);
+  if (audio_senders_.empty())
+    return;
+
+  auto it = audio_senders_.begin();
+  while (++it != audio_senders_.end()) {
+    auto audio_frame_copy = std::make_unique<AudioFrame>();
+    audio_frame_copy->CopyFrom(*audio_frame);
+    (*it)->SendAudioData(std::move(audio_frame_copy));
+  }
+  // Send the original frame to the first stream w/o copying.
+  (*audio_senders_.begin())->SendAudioData(std::move(audio_frame));
+}
+
+// Mix all received streams, feed the result to the AudioProcessing module, then
+// resample the result to the requested output rate.
+int32_t AudioTransportImpl::NeedMorePlayData(const size_t nSamples,
+                                             const size_t nBytesPerSample,
+                                             const size_t nChannels,
+                                             const uint32_t samplesPerSec,
+                                             void* audioSamples,
+                                             size_t& nSamplesOut,
+                                             int64_t* elapsed_time_ms,
+                                             int64_t* ntp_time_ms) {
+  TRACE_EVENT0("webrtc", "AudioTransportImpl::SendProcessedData");
+  RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
+  RTC_DCHECK_GE(nChannels, 1);
+  RTC_DCHECK_LE(nChannels, 2);
+  RTC_DCHECK_GE(
+      samplesPerSec,
+      static_cast<uint32_t>(AudioProcessing::NativeRate::kSampleRate8kHz));
+
+  // 100 = 1 second / data duration (10 ms).
+  RTC_DCHECK_EQ(nSamples * 100, samplesPerSec);
+  RTC_DCHECK_LE(nBytesPerSample * nSamples * nChannels,
+                AudioFrame::kMaxDataSizeBytes);
+
+  mixer_->Mix(nChannels, &mixed_frame_);
+  *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
+  *ntp_time_ms = mixed_frame_.ntp_time_ms_;
+
+  if (audio_processing_) {
+    const auto error =
+        ProcessReverseAudioFrame(audio_processing_, &mixed_frame_);
+    RTC_DCHECK_EQ(error, AudioProcessing::kNoError);
+  }
+
+  nSamplesOut = Resample(mixed_frame_, samplesPerSec, &render_resampler_,
+                         static_cast<int16_t*>(audioSamples));
+  RTC_DCHECK_EQ(nSamplesOut, nChannels * nSamples);
+  return 0;
+}
+
+// Used by Chromium - same as NeedMorePlayData() but because Chrome has its
+// own APM instance, does not call audio_processing_->ProcessReverseStream().
+void AudioTransportImpl::PullRenderData(int bits_per_sample,
+                                        int sample_rate,
+                                        size_t number_of_channels,
+                                        size_t number_of_frames,
+                                        void* audio_data,
+                                        int64_t* elapsed_time_ms,
+                                        int64_t* ntp_time_ms) {
+  TRACE_EVENT2("webrtc", "AudioTransportImpl::PullRenderData", "sample_rate",
+               sample_rate, "number_of_frames", number_of_frames);
+  RTC_DCHECK_EQ(bits_per_sample, 16);
+  RTC_DCHECK_GE(number_of_channels, 1);
+  RTC_DCHECK_GE(sample_rate, AudioProcessing::NativeRate::kSampleRate8kHz);
+
+  // 100 = 1 second / data duration (10 ms).
+  RTC_DCHECK_EQ(number_of_frames * 100, sample_rate);
+
+  // 8 = bits per byte.
+  RTC_DCHECK_LE(bits_per_sample / 8 * number_of_frames * number_of_channels,
+                AudioFrame::kMaxDataSizeBytes);
+  mixer_->Mix(number_of_channels, &mixed_frame_);
+  *elapsed_time_ms = mixed_frame_.elapsed_time_ms_;
+  *ntp_time_ms = mixed_frame_.ntp_time_ms_;
+
+  auto output_samples = Resample(mixed_frame_, sample_rate, &render_resampler_,
+                                 static_cast<int16_t*>(audio_data));
+  RTC_DCHECK_EQ(output_samples, number_of_channels * number_of_frames);
+}
+
+void AudioTransportImpl::UpdateAudioSenders(std::vector<AudioSender*> senders,
+                                            int send_sample_rate_hz,
+                                            size_t send_num_channels) {
+  MutexLock lock(&capture_lock_);
+  audio_senders_ = std::move(senders);
+  send_sample_rate_hz_ = send_sample_rate_hz;
+  send_num_channels_ = send_num_channels;
+}
+
+void AudioTransportImpl::SetStereoChannelSwapping(bool enable) {
+  MutexLock lock(&capture_lock_);
+  swap_stereo_channels_ = enable;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/audio_transport_impl.h b/third_party/libwebrtc/audio/audio_transport_impl.h
new file mode 100644
index 0000000000..24b09d2140
--- /dev/null
+++ b/third_party/libwebrtc/audio/audio_transport_impl.h
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_AUDIO_TRANSPORT_IMPL_H_
+#define AUDIO_AUDIO_TRANSPORT_IMPL_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio/audio_mixer.h"
+#include "api/scoped_refptr.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "modules/async_audio_processing/async_audio_processing.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class AudioSender;
+
+class AudioTransportImpl : public AudioTransport {
+ public:
+  AudioTransportImpl(
+      AudioMixer* mixer,
+      AudioProcessing* audio_processing,
+      AsyncAudioProcessing::Factory* async_audio_processing_factory);
+
+  AudioTransportImpl() = delete;
+  AudioTransportImpl(const AudioTransportImpl&) = delete;
+  AudioTransportImpl& operator=(const AudioTransportImpl&) = delete;
+
+  ~AudioTransportImpl() override;
+
+  // TODO(bugs.webrtc.org/13620) Deprecate this function
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  size_t nSamples,
+                                  size_t nBytesPerSample,
+                                  size_t nChannels,
+                                  uint32_t samplesPerSec,
+                                  uint32_t totalDelayMS,
+                                  int32_t clockDrift,
+                                  uint32_t currentMicLevel,
+                                  bool keyPressed,
+                                  uint32_t& newMicLevel) override;
+
+  int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t nSamples,
+      size_t nBytesPerSample,
+      size_t nChannels,
+      uint32_t samplesPerSec,
+      uint32_t totalDelayMS,
+      int32_t clockDrift,
+      uint32_t currentMicLevel,
+      bool keyPressed,
+      uint32_t& newMicLevel,
+      absl::optional<int64_t> estimated_capture_time_ns) override;
+
+  int32_t NeedMorePlayData(size_t nSamples,
+                           size_t nBytesPerSample,
+                           size_t nChannels,
+                           uint32_t samplesPerSec,
+                           void* audioSamples,
+                           size_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override;
+
+  void PullRenderData(int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      void* audio_data,
+                      int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms) override;
+
+  void UpdateAudioSenders(std::vector<AudioSender*> senders,
+                          int send_sample_rate_hz,
+                          size_t send_num_channels);
+  void SetStereoChannelSwapping(bool enable);
+
+ private:
+  void SendProcessedData(std::unique_ptr<AudioFrame> audio_frame);
+
+  // Shared.
+  AudioProcessing* audio_processing_ = nullptr;
+
+  // Capture side.
+
+  // Thread-safe.
+  const std::unique_ptr<AsyncAudioProcessing> async_audio_processing_;
+
+  mutable Mutex capture_lock_;
+  std::vector<AudioSender*> audio_senders_ RTC_GUARDED_BY(capture_lock_);
+  int send_sample_rate_hz_ RTC_GUARDED_BY(capture_lock_) = 8000;
+  size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;
+  bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
+  PushResampler<int16_t> capture_resampler_;
+
+  // Render side.
+
+  rtc::scoped_refptr<AudioMixer> mixer_;
+  AudioFrame mixed_frame_;
+  // Converts mixed audio to the audio device output rate.
+  PushResampler<int16_t> render_resampler_;
+};
+}  // namespace webrtc
+
+#endif  // AUDIO_AUDIO_TRANSPORT_IMPL_H_
diff --git a/third_party/libwebrtc/audio/channel_receive.cc b/third_party/libwebrtc/audio/channel_receive.cc
new file mode 100644
index 0000000000..c714b1dd4d
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_receive.cc
@@ -0,0 +1,1129 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/channel_receive.h"
+
+#include <algorithm>
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/frame_transformer_interface.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/time_delta.h"
+#include "audio/audio_level.h"
+#include "audio/channel_receive_frame_transformer_delegate.h"
+#include "audio/channel_send.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
+#include "logging/rtc_event_log/events/rtc_event_neteq_set_minimum_delay.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "modules/rtp_rtcp/source/absolute_capture_time_interpolator.h"
+#include "modules/rtp_rtcp/source/capture_clock_offset_updater.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/metrics.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+namespace voe {
+
+namespace {
+
+constexpr double kAudioSampleDurationSeconds = 0.01;
+
+// Video Sync.
+constexpr int kVoiceEngineMinMinPlayoutDelayMs = 0;
+constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
+
+acm2::AcmReceiver::Config AcmConfig(
+    NetEqFactory* neteq_factory,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+    absl::optional<AudioCodecPairId> codec_pair_id,
+    size_t jitter_buffer_max_packets,
+    bool jitter_buffer_fast_playout,
+    int jitter_buffer_min_delay_ms) {
+  acm2::AcmReceiver::Config acm_config;
+  acm_config.neteq_factory = neteq_factory;
+  acm_config.decoder_factory = decoder_factory;
+  acm_config.neteq_config.codec_pair_id = codec_pair_id;
+  acm_config.neteq_config.max_packets_in_buffer = jitter_buffer_max_packets;
+  acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout;
+  acm_config.neteq_config.enable_muted_state = true;
+  acm_config.neteq_config.min_delay_ms = jitter_buffer_min_delay_ms;
+
+  return acm_config;
+}
+
+class ChannelReceive : public ChannelReceiveInterface,
+                       public RtcpPacketTypeCounterObserver {
+ public:
+  // Used for receive streams.
+  ChannelReceive(
+      Clock* clock,
+      NetEqFactory* neteq_factory,
+      AudioDeviceModule* audio_device_module,
+      Transport* rtcp_send_transport,
+      RtcEventLog* rtc_event_log,
+      uint32_t local_ssrc,
+      uint32_t remote_ssrc,
+      size_t jitter_buffer_max_packets,
+      bool jitter_buffer_fast_playout,
+      int jitter_buffer_min_delay_ms,
+      bool enable_non_sender_rtt,
+      rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
+      const webrtc::CryptoOptions& crypto_options,
+      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+      RtcpEventObserver* rtcp_event_observer);
+  ~ChannelReceive() override;
+
+  void SetSink(AudioSinkInterface* sink) override;
+
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
+
+  // API methods
+
+  void StartPlayout() override;
+  void StopPlayout() override;
+
+  // Codecs
+  absl::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec()
+      const override;
+
+  void ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
+
+  // RtpPacketSinkInterface.
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
+
+  // Muting, Volume and Level.
+  void SetChannelOutputVolumeScaling(float scaling) override;
+  int GetSpeechOutputLevelFullRange() const override;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec:
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+  double GetTotalOutputEnergy() const override;
+  double GetTotalOutputDuration() const override;
+
+  // Stats.
+  NetworkStatistics GetNetworkStatistics(
+      bool get_and_clear_legacy_stats) const override;
+  AudioDecodingCallStats GetDecodingCallStatistics() const override;
+
+  // Audio+Video Sync.
+  uint32_t GetDelayEstimate() const override;
+  bool SetMinimumPlayoutDelay(int delayMs) override;
+  bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+                              int64_t* time_ms) const override;
+  void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
+                                         int64_t time_ms) override;
+  absl::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
+      int64_t now_ms) const override;
+
+  // Audio quality.
+  bool SetBaseMinimumPlayoutDelayMs(int delay_ms) override;
+  int GetBaseMinimumPlayoutDelayMs() const override;
+
+  // Produces the transport-related timestamps; current_delay_ms is left unset.
+  absl::optional<Syncable::Info> GetSyncInfo() const override;
+
+  void RegisterReceiverCongestionControlObjects(
+      PacketRouter* packet_router) override;
+  void ResetReceiverCongestionControlObjects() override;
+
+  CallReceiveStatistics GetRTCPStatistics() const override;
+  void SetNACKStatus(bool enable, int maxNumberOfPackets) override;
+  void SetNonSenderRttMeasurement(bool enabled) override;
+
+  AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
+      int sample_rate_hz,
+      AudioFrame* audio_frame) override;
+
+  int PreferredSampleRate() const override;
+
+  void SetSourceTracker(SourceTracker* source_tracker) override;
+
+  // Associate to a send channel.
+  // Used for obtaining RTT for a receive-only channel.
+  void SetAssociatedSendChannel(const ChannelSendInterface* channel) override;
+
+  // Sets a frame transformer between the depacketizer and the decoder, to
+  // transform the received frames before decoding them.
+  void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+
+  void SetFrameDecryptor(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>
+                             frame_decryptor) override;
+
+  void OnLocalSsrcChange(uint32_t local_ssrc) override;
+  uint32_t GetLocalSsrc() const override;
+
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) override;
+
+ private:
+  void ReceivePacket(const uint8_t* packet,
+                     size_t packet_length,
+                     const RTPHeader& header)
+      RTC_RUN_ON(worker_thread_checker_);
+  int ResendPackets(const uint16_t* sequence_numbers, int length);
+  void UpdatePlayoutTimestamp(bool rtcp, int64_t now_ms)
+      RTC_RUN_ON(worker_thread_checker_);
+
+  int GetRtpTimestampRateHz() const;
+
+  void OnReceivedPayloadData(rtc::ArrayView<const uint8_t> payload,
+                             const RTPHeader& rtpHeader)
+      RTC_RUN_ON(worker_thread_checker_);
+
+  void InitFrameTransformerDelegate(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      RTC_RUN_ON(worker_thread_checker_);
+
+  // Thread checkers document and lock usage of some methods to specific threads
+  // we know about. The goal is to eventually split up voe::ChannelReceive into
+  // parts with single-threaded semantics, and thereby reduce the need for
+  // locks.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker network_thread_checker_;
+
+  TaskQueueBase* const worker_thread_;
+  ScopedTaskSafety worker_safety_;
+
+  // Methods accessed from audio and video threads are checked for sequential-
+  // only access. We don't necessarily own and control these threads, so thread
+  // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
+  // audio thread to another, but access is still sequential.
+  rtc::RaceChecker audio_thread_race_checker_;
+  Mutex callback_mutex_;
+  Mutex volume_settings_mutex_;
+
+  bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
+
+  RtcEventLog* const event_log_;
+
+  // Indexed by payload type.
+  std::map<uint8_t, int> payload_type_frequencies_;
+
+  std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
+  const uint32_t remote_ssrc_;
+  SourceTracker* source_tracker_ = nullptr;
+
+  // Info for GetSyncInfo is updated on network or worker thread, and queried on
+  // the worker thread.
+  absl::optional<uint32_t> last_received_rtp_timestamp_
+      RTC_GUARDED_BY(&worker_thread_checker_);
+  absl::optional<int64_t> last_received_rtp_system_time_ms_
+      RTC_GUARDED_BY(&worker_thread_checker_);
+
+  // The AcmReceiver is thread safe, using its own lock.
+  acm2::AcmReceiver acm_receiver_;
+  AudioSinkInterface* audio_sink_ = nullptr;
+  AudioLevel _outputAudioLevel;
+
+  Clock* const clock_;
+  RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
+
+  // Timestamp of the audio pulled from NetEq.
+  absl::optional<uint32_t> jitter_buffer_playout_timestamp_;
+
+  uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(worker_thread_checker_);
+  absl::optional<int64_t> playout_timestamp_rtp_time_ms_
+      RTC_GUARDED_BY(worker_thread_checker_);
+  uint32_t playout_delay_ms_ RTC_GUARDED_BY(worker_thread_checker_);
+  absl::optional<int64_t> playout_timestamp_ntp_
+      RTC_GUARDED_BY(worker_thread_checker_);
+  absl::optional<int64_t> playout_timestamp_ntp_time_ms_
+      RTC_GUARDED_BY(worker_thread_checker_);
+
+  mutable Mutex ts_stats_lock_;
+
+  webrtc::RtpTimestampUnwrapper rtp_ts_wraparound_handler_;
+  // The rtp timestamp of the first played out audio frame.
+  int64_t capture_start_rtp_time_stamp_;
+  // The capture ntp time (in local timebase) of the first played out audio
+  // frame.
+  int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_);
+
+  AudioDeviceModule* _audioDeviceModulePtr;
+  float _outputGain RTC_GUARDED_BY(volume_settings_mutex_);
+
+  const ChannelSendInterface* associated_send_channel_
+      RTC_GUARDED_BY(network_thread_checker_);
+
+  PacketRouter* packet_router_ = nullptr;
+
+  SequenceChecker construction_thread_;
+
+  // E2EE Audio Frame Decryption
+  rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_
+      RTC_GUARDED_BY(worker_thread_checker_);
+  webrtc::CryptoOptions crypto_options_;
+
+  webrtc::AbsoluteCaptureTimeInterpolator absolute_capture_time_interpolator_
+      RTC_GUARDED_BY(worker_thread_checker_);
+
+  webrtc::CaptureClockOffsetUpdater capture_clock_offset_updater_
+      RTC_GUARDED_BY(ts_stats_lock_);
+
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate>
+      frame_transformer_delegate_;
+
+  // Counter that's used to control the frequency of reporting histograms
+  // from the `GetAudioFrameWithInfo` callback.
+  int audio_frame_interval_count_ RTC_GUARDED_BY(audio_thread_race_checker_) =
+      0;
+  // Controls how many callbacks we let pass by before reporting callback stats.
+  // A value of 100 means 100 callbacks, each one of which represents 10ms worth
+  // of data, so the stats reporting frequency will be 1Hz (modulo failures).
+  constexpr static int kHistogramReportingInterval = 100;
+
+  mutable Mutex rtcp_counter_mutex_;
+  RtcpPacketTypeCounter rtcp_packet_type_counter_
+      RTC_GUARDED_BY(rtcp_counter_mutex_);
+};
+
+void ChannelReceive::OnReceivedPayloadData(
+    rtc::ArrayView<const uint8_t> payload,
+    const RTPHeader& rtpHeader) {
+  if (!playing_) {
+    // Avoid inserting into NetEQ when we are not playing. Count the
+    // packet as discarded.
+
+    // If we have a source_tracker_, tell it that the frame has been
+    // "delivered". Normally, this happens in AudioReceiveStreamInterface when
+    // audio frames are pulled out, but when playout is muted, nothing is
+    // pulling frames. The downside of this approach is that frames delivered
+    // this way won't be delayed for playout, and therefore will be
+    // unsynchronized with (a) audio delay when playing and (b) any audio/video
+    // synchronization. But the alternative is that muting playout also stops
+    // the SourceTracker from updating RtpSource information.
+    if (source_tracker_) {
+      RtpPacketInfos::vector_type packet_vector = {
+          RtpPacketInfo(rtpHeader, clock_->CurrentTime())};
+      source_tracker_->OnFrameDelivered(RtpPacketInfos(packet_vector));
+    }
+
+    return;
+  }
+
+  // Push the incoming payload (parsed and ready for decoding) into the ACM
+  if (acm_receiver_.InsertPacket(rtpHeader, payload) != 0) {
+    RTC_DLOG(LS_ERROR) << "ChannelReceive::OnReceivedPayloadData() unable to "
+                          "push data to the ACM";
+    return;
+  }
+
+  TimeDelta round_trip_time = rtp_rtcp_->LastRtt().value_or(TimeDelta::Zero());
+
+  std::vector<uint16_t> nack_list =
+      acm_receiver_.GetNackList(round_trip_time.ms());
+  if (!nack_list.empty()) {
+    // Can't use nack_list.data() since it's not supported by all
+    // compilers.
+    ResendPackets(&(nack_list[0]), static_cast<int>(nack_list.size()));
+  }
+}
+
+void ChannelReceive::InitFrameTransformerDelegate(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK(frame_transformer);
+  RTC_DCHECK(!frame_transformer_delegate_);
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  // Pass a callback to ChannelReceive::OnReceivedPayloadData, to be called by
+  // the delegate to receive transformed audio.
+  ChannelReceiveFrameTransformerDelegate::ReceiveFrameCallback
+      receive_audio_callback = [this](rtc::ArrayView<const uint8_t> packet,
+                                      const RTPHeader& header) {
+        RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+        OnReceivedPayloadData(packet, header);
+      };
+  frame_transformer_delegate_ =
+      rtc::make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
+          std::move(receive_audio_callback), std::move(frame_transformer),
+          worker_thread_);
+  frame_transformer_delegate_->Init();
+}
+
+AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
+    int sample_rate_hz,
+    AudioFrame* audio_frame) {
+  TRACE_EVENT_BEGIN1("webrtc", "ChannelReceive::GetAudioFrameWithInfo",
+                     "sample_rate_hz", sample_rate_hz);
+  RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
+  audio_frame->sample_rate_hz_ = sample_rate_hz;
+
+  event_log_->Log(std::make_unique<RtcEventAudioPlayout>(remote_ssrc_));
+
+  // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
+  bool muted;
+  if (acm_receiver_.GetAudio(audio_frame->sample_rate_hz_, audio_frame,
+                             &muted) == -1) {
+    RTC_DLOG(LS_ERROR)
+        << "ChannelReceive::GetAudioFrame() PlayoutData10Ms() failed!";
+    // In all likelihood, the audio in this frame is garbage. We return an
+    // error so that the audio mixer module doesn't add it to the mix. As
+    // a result, it won't be played out and the actions skipped here are
+    // irrelevant.
+
+    TRACE_EVENT_END1("webrtc", "ChannelReceive::GetAudioFrameWithInfo", "error",
+                     1);
+    return AudioMixer::Source::AudioFrameInfo::kError;
+  }
+
+  if (muted) {
+    // TODO(henrik.lundin): We should be able to do better than this. But we
+    // will have to go through all the cases below where the audio samples may
+    // be used, and handle the muted case in some way.
+    AudioFrameOperations::Mute(audio_frame);
+  }
+
+  {
+    // Pass the audio buffers to an optional sink callback, before applying
+    // scaling/panning, as that applies to the mix operation.
+    // External recipients of the audio (e.g. via AudioTrack), will do their
+    // own mixing/dynamic processing.
+    MutexLock lock(&callback_mutex_);
+    if (audio_sink_) {
+      AudioSinkInterface::Data data(
+          audio_frame->data(), audio_frame->samples_per_channel_,
+          audio_frame->sample_rate_hz_, audio_frame->num_channels_,
+          audio_frame->timestamp_);
+      audio_sink_->OnData(data);
+    }
+  }
+
+  float output_gain = 1.0f;
+  {
+    MutexLock lock(&volume_settings_mutex_);
+    output_gain = _outputGain;
+  }
+
+  // Output volume scaling
+  if (output_gain < 0.99f || output_gain > 1.01f) {
+    // TODO(solenberg): Combine with mute state - this can cause clicks!
+    AudioFrameOperations::ScaleWithSat(output_gain, audio_frame);
+  }
+
+  // Measure audio level (0-9)
+  // TODO(henrik.lundin) Use the `muted` information here too.
+  // TODO(deadbeef): Use RmsLevel for `_outputAudioLevel` (see
+  // https://crbug.com/webrtc/7517).
+  _outputAudioLevel.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds);
+
+  if (capture_start_rtp_time_stamp_ < 0 && audio_frame->timestamp_ != 0) {
+    // The first frame with a valid rtp timestamp.
+    capture_start_rtp_time_stamp_ = audio_frame->timestamp_;
+  }
+
+  if (capture_start_rtp_time_stamp_ >= 0) {
+    // audio_frame.timestamp_ should be valid from now on.
+    // Compute elapsed time.
+    int64_t unwrap_timestamp =
+        rtp_ts_wraparound_handler_.Unwrap(audio_frame->timestamp_);
+    audio_frame->elapsed_time_ms_ =
+        (unwrap_timestamp - capture_start_rtp_time_stamp_) /
+        (GetRtpTimestampRateHz() / 1000);
+
+    {
+      MutexLock lock(&ts_stats_lock_);
+      // Compute ntp time.
+      audio_frame->ntp_time_ms_ =
+          ntp_estimator_.Estimate(audio_frame->timestamp_);
+      // `ntp_time_ms_` won't be valid until at least 2 RTCP SRs are received.
+      if (audio_frame->ntp_time_ms_ > 0) {
+        // Compute `capture_start_ntp_time_ms_` so that
+        // `capture_start_ntp_time_ms_` + `elapsed_time_ms_` == `ntp_time_ms_`
+        capture_start_ntp_time_ms_ =
+            audio_frame->ntp_time_ms_ - audio_frame->elapsed_time_ms_;
+      }
+    }
+  }
+
+  // Fill in local capture clock offset in `audio_frame->packet_infos_`.
+  RtpPacketInfos::vector_type packet_infos;
+  for (auto& packet_info : audio_frame->packet_infos_) {
+    RtpPacketInfo new_packet_info(packet_info);
+    if (packet_info.absolute_capture_time().has_value()) {
+      MutexLock lock(&ts_stats_lock_);
+      new_packet_info.set_local_capture_clock_offset(
+          capture_clock_offset_updater_.ConvertsToTimeDela(
+              capture_clock_offset_updater_.AdjustEstimatedCaptureClockOffset(
+                  packet_info.absolute_capture_time()
+                      ->estimated_capture_clock_offset)));
+    }
+    packet_infos.push_back(std::move(new_packet_info));
+  }
+  audio_frame->packet_infos_ = RtpPacketInfos(packet_infos);
+
+  ++audio_frame_interval_count_;
+  if (audio_frame_interval_count_ >= kHistogramReportingInterval) {
+    audio_frame_interval_count_ = 0;
+    worker_thread_->PostTask(SafeTask(worker_safety_.flag(), [this]() {
+      RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+      RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.TargetJitterBufferDelayMs",
+                                acm_receiver_.TargetDelayMs());
+      const int jitter_buffer_delay = acm_receiver_.FilteredCurrentDelayMs();
+      RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDelayEstimateMs",
+                                jitter_buffer_delay + playout_delay_ms_);
+      RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverJitterBufferDelayMs",
+                                jitter_buffer_delay);
+      RTC_HISTOGRAM_COUNTS_1000("WebRTC.Audio.ReceiverDeviceDelayMs",
+                                playout_delay_ms_);
+    }));
+  }
+
+  TRACE_EVENT_END2("webrtc", "ChannelReceive::GetAudioFrameWithInfo", "gain",
+                   output_gain, "muted", muted);
+  return muted ? AudioMixer::Source::AudioFrameInfo::kMuted
+               : AudioMixer::Source::AudioFrameInfo::kNormal;
+}
+
+int ChannelReceive::PreferredSampleRate() const {
+  RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
+  // Return the bigger of playout and receive frequency in the ACM.
+  return std::max(acm_receiver_.last_packet_sample_rate_hz().value_or(0),
+                  acm_receiver_.last_output_sample_rate_hz());
+}
+
+void ChannelReceive::SetSourceTracker(SourceTracker* source_tracker) {
+  source_tracker_ = source_tracker;
+}
+
+ChannelReceive::ChannelReceive(
+    Clock* clock,
+    NetEqFactory* neteq_factory,
+    AudioDeviceModule* audio_device_module,
+    Transport* rtcp_send_transport,
+    RtcEventLog* rtc_event_log,
+    uint32_t local_ssrc,
+    uint32_t remote_ssrc,
+    size_t jitter_buffer_max_packets,
+    bool jitter_buffer_fast_playout,
+    int jitter_buffer_min_delay_ms,
+    bool enable_non_sender_rtt,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+    absl::optional<AudioCodecPairId> codec_pair_id,
+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    RtcpEventObserver* rtcp_event_observer)
+    : worker_thread_(TaskQueueBase::Current()),
+      event_log_(rtc_event_log),
+      rtp_receive_statistics_(ReceiveStatistics::Create(clock)),
+      remote_ssrc_(remote_ssrc),
+      acm_receiver_(AcmConfig(neteq_factory,
+                              decoder_factory,
+                              codec_pair_id,
+                              jitter_buffer_max_packets,
+                              jitter_buffer_fast_playout,
+                              jitter_buffer_min_delay_ms)),
+      _outputAudioLevel(),
+      clock_(clock),
+      ntp_estimator_(clock),
+      playout_timestamp_rtp_(0),
+      playout_delay_ms_(0),
+      capture_start_rtp_time_stamp_(-1),
+      capture_start_ntp_time_ms_(-1),
+      _audioDeviceModulePtr(audio_device_module),
+      _outputGain(1.0f),
+      associated_send_channel_(nullptr),
+      frame_decryptor_(frame_decryptor),
+      crypto_options_(crypto_options),
+      absolute_capture_time_interpolator_(clock) {
+  RTC_DCHECK(audio_device_module);
+
+  network_thread_checker_.Detach();
+
+  acm_receiver_.ResetInitialDelay();
+  acm_receiver_.SetMinimumDelay(0);
+  acm_receiver_.SetMaximumDelay(0);
+  acm_receiver_.FlushBuffers();
+
+  _outputAudioLevel.ResetLevelFullRange();
+
+  rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc_, true);
+  RtpRtcpInterface::Configuration configuration;
+  configuration.clock = clock;
+  configuration.audio = true;
+  configuration.receiver_only = true;
+  configuration.outgoing_transport = rtcp_send_transport;
+  configuration.receive_statistics = rtp_receive_statistics_.get();
+  configuration.event_log = event_log_;
+  configuration.local_media_ssrc = local_ssrc;
+  configuration.rtcp_packet_type_counter_observer = this;
+  configuration.non_sender_rtt_measurement = enable_non_sender_rtt;
+  configuration.rtcp_event_observer = rtcp_event_observer;
+
+  if (frame_transformer)
+    InitFrameTransformerDelegate(std::move(frame_transformer));
+
+  rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration);
+  rtp_rtcp_->SetRemoteSSRC(remote_ssrc_);
+
+  // Ensure that RTCP is enabled for the created channel.
+  rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
+}
+
+ChannelReceive::~ChannelReceive() {
+  RTC_DCHECK_RUN_ON(&construction_thread_);
+
+  // Resets the delegate's callback to ChannelReceive::OnReceivedPayloadData.
+  if (frame_transformer_delegate_)
+    frame_transformer_delegate_->Reset();
+
+  StopPlayout();
+}
+
+void ChannelReceive::SetSink(AudioSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  MutexLock lock(&callback_mutex_);
+  audio_sink_ = sink;
+}
+
+void ChannelReceive::StartPlayout() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  playing_ = true;
+}
+
+void ChannelReceive::StopPlayout() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  playing_ = false;
+  _outputAudioLevel.ResetLevelFullRange();
+  acm_receiver_.FlushBuffers();
+}
+
+absl::optional<std::pair<int, SdpAudioFormat>> ChannelReceive::GetReceiveCodec()
+    const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return acm_receiver_.LastDecoder();
+}
+
+void ChannelReceive::SetReceiveCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  for (const auto& kv : codecs) {
+    RTC_DCHECK_GE(kv.second.clockrate_hz, 1000);
+    payload_type_frequencies_[kv.first] = kv.second.clockrate_hz;
+  }
+  acm_receiver_.SetCodecs(codecs);
+}
+
+void ChannelReceive::OnRtpPacket(const RtpPacketReceived& packet) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // TODO(bugs.webrtc.org/11993): Expect to be called exclusively on the
+  // network thread. Once that's done, the same applies to
+  // UpdatePlayoutTimestamp and
+  int64_t now_ms = rtc::TimeMillis();
+
+  last_received_rtp_timestamp_ = packet.Timestamp();
+  last_received_rtp_system_time_ms_ = now_ms;
+
+  // Store playout timestamp for the received RTP packet
+  UpdatePlayoutTimestamp(false, now_ms);
+
+  const auto& it = payload_type_frequencies_.find(packet.PayloadType());
+  if (it == payload_type_frequencies_.end())
+    return;
+  // TODO(bugs.webrtc.org/7135): Set payload_type_frequency earlier, when packet
+  // is parsed.
+  RtpPacketReceived packet_copy(packet);
+  packet_copy.set_payload_type_frequency(it->second);
+
+  rtp_receive_statistics_->OnRtpPacket(packet_copy);
+
+  RTPHeader header;
+  packet_copy.GetHeader(&header);
+
+  // Interpolates absolute capture timestamp RTP header extension.
+  header.extension.absolute_capture_time =
+      absolute_capture_time_interpolator_.OnReceivePacket(
+          AbsoluteCaptureTimeInterpolator::GetSource(header.ssrc,
+                                                     header.arrOfCSRCs),
+          header.timestamp,
+          rtc::saturated_cast<uint32_t>(packet_copy.payload_type_frequency()),
+          header.extension.absolute_capture_time);
+
+  ReceivePacket(packet_copy.data(), packet_copy.size(), header);
+}
+
+void ChannelReceive::ReceivePacket(const uint8_t* packet,
+                                   size_t packet_length,
+                                   const RTPHeader& header) {
+  const uint8_t* payload = packet + header.headerLength;
+  RTC_DCHECK_GE(packet_length, header.headerLength);
+  size_t payload_length = packet_length - header.headerLength;
+
+  size_t payload_data_length = payload_length - header.paddingLength;
+
+  // E2EE Custom Audio Frame Decryption (This is optional).
+  // Keep this buffer around for the lifetime of the OnReceivedPayloadData call.
+  rtc::Buffer decrypted_audio_payload;
+  if (frame_decryptor_ != nullptr) {
+    const size_t max_plaintext_size = frame_decryptor_->GetMaxPlaintextByteSize(
+        cricket::MEDIA_TYPE_AUDIO, payload_length);
+    decrypted_audio_payload.SetSize(max_plaintext_size);
+
+    const std::vector<uint32_t> csrcs(header.arrOfCSRCs,
+                                      header.arrOfCSRCs + header.numCSRCs);
+    const FrameDecryptorInterface::Result decrypt_result =
+        frame_decryptor_->Decrypt(
+            cricket::MEDIA_TYPE_AUDIO, csrcs,
+            /*additional_data=*/nullptr,
+            rtc::ArrayView<const uint8_t>(payload, payload_data_length),
+            decrypted_audio_payload);
+
+    if (decrypt_result.IsOk()) {
+      decrypted_audio_payload.SetSize(decrypt_result.bytes_written);
+    } else {
+      // Interpret failures as a silent frame.
+      decrypted_audio_payload.SetSize(0);
+    }
+
+    payload = decrypted_audio_payload.data();
+    payload_data_length = decrypted_audio_payload.size();
+  } else if (crypto_options_.sframe.require_frame_encryption) {
+    RTC_DLOG(LS_ERROR)
+        << "FrameDecryptor required but not set, dropping packet";
+    payload_data_length = 0;
+  }
+
+  rtc::ArrayView<const uint8_t> payload_data(payload, payload_data_length);
+  if (frame_transformer_delegate_) {
+    // Asynchronously transform the received payload. After the payload is
+    // transformed, the delegate will call OnReceivedPayloadData to handle it.
+    frame_transformer_delegate_->Transform(payload_data, header, remote_ssrc_);
+  } else {
+    OnReceivedPayloadData(payload_data, header);
+  }
+}
+
+void ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // TODO(bugs.webrtc.org/11993): Expect to be called exclusively on the
+  // network thread.
+
+  // Store playout timestamp for the received RTCP packet
+  UpdatePlayoutTimestamp(true, rtc::TimeMillis());
+
+  // Deliver RTCP packet to RTP/RTCP module for parsing
+  rtp_rtcp_->IncomingRtcpPacket(rtc::MakeArrayView(data, length));
+
+  absl::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
+  if (!rtt.has_value()) {
+    // Waiting for valid RTT.
+    return;
+  }
+
+  absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+      rtp_rtcp_->GetSenderReportStats();
+  if (!last_sr.has_value()) {
+    // Waiting for RTCP.
+    return;
+  }
+
+  {
+    MutexLock lock(&ts_stats_lock_);
+    ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_timestamp,
+                                       last_sr->last_remote_rtp_timestamp);
+    absl::optional<int64_t> remote_to_local_clock_offset =
+        ntp_estimator_.EstimateRemoteToLocalClockOffset();
+    if (remote_to_local_clock_offset.has_value()) {
+      capture_clock_offset_updater_.SetRemoteToLocalClockOffset(
+          *remote_to_local_clock_offset);
+    }
+  }
+}
+
+int ChannelReceive::GetSpeechOutputLevelFullRange() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return _outputAudioLevel.LevelFullRange();
+}
+
+double ChannelReceive::GetTotalOutputEnergy() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return _outputAudioLevel.TotalEnergy();
+}
+
+double ChannelReceive::GetTotalOutputDuration() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return _outputAudioLevel.TotalDuration();
+}
+
+void ChannelReceive::SetChannelOutputVolumeScaling(float scaling) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  MutexLock lock(&volume_settings_mutex_);
+  _outputGain = scaling;
+}
+
+void ChannelReceive::RegisterReceiverCongestionControlObjects(
+    PacketRouter* packet_router) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK(packet_router);
+  RTC_DCHECK(!packet_router_);
+  constexpr bool remb_candidate = false;
+  packet_router->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
+  packet_router_ = packet_router;
+}
+
+void ChannelReceive::ResetReceiverCongestionControlObjects() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK(packet_router_);
+  packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get());
+  packet_router_ = nullptr;
+}
+
+CallReceiveStatistics ChannelReceive::GetRTCPStatistics() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  CallReceiveStatistics stats;
+
+  // The jitter statistics is updated for each received RTP packet and is based
+  // on received packets.
+  RtpReceiveStats rtp_stats;
+  StreamStatistician* statistician =
+      rtp_receive_statistics_->GetStatistician(remote_ssrc_);
+  if (statistician) {
+    rtp_stats = statistician->GetStats();
+  }
+
+  stats.cumulativeLost = rtp_stats.packets_lost;
+  stats.jitterSamples = rtp_stats.jitter;
+
+  // Data counters.
+  if (statistician) {
+    stats.payload_bytes_received = rtp_stats.packet_counter.payload_bytes;
+
+    stats.header_and_padding_bytes_received =
+        rtp_stats.packet_counter.header_bytes +
+        rtp_stats.packet_counter.padding_bytes;
+    stats.packetsReceived = rtp_stats.packet_counter.packets;
+    stats.last_packet_received = rtp_stats.last_packet_received;
+  } else {
+    stats.payload_bytes_received = 0;
+    stats.header_and_padding_bytes_received = 0;
+    stats.packetsReceived = 0;
+    stats.last_packet_received = absl::nullopt;
+  }
+
+  {
+    MutexLock lock(&rtcp_counter_mutex_);
+    stats.nacks_sent = rtcp_packet_type_counter_.nack_packets;
+  }
+
+  // Timestamps.
+  {
+    MutexLock lock(&ts_stats_lock_);
+    stats.capture_start_ntp_time_ms_ = capture_start_ntp_time_ms_;
+  }
+
+  absl::optional<RtpRtcpInterface::SenderReportStats> rtcp_sr_stats =
+      rtp_rtcp_->GetSenderReportStats();
+  if (rtcp_sr_stats.has_value()) {
+    stats.last_sender_report_timestamp_ms =
+        rtcp_sr_stats->last_arrival_timestamp.ToMs() -
+        rtc::kNtpJan1970Millisecs;
+    stats.last_sender_report_remote_timestamp_ms =
+        rtcp_sr_stats->last_remote_timestamp.ToMs() - rtc::kNtpJan1970Millisecs;
+    stats.sender_reports_packets_sent = rtcp_sr_stats->packets_sent;
+    stats.sender_reports_bytes_sent = rtcp_sr_stats->bytes_sent;
+    stats.sender_reports_reports_count = rtcp_sr_stats->reports_count;
+  }
+
+  absl::optional<RtpRtcpInterface::NonSenderRttStats> non_sender_rtt_stats =
+      rtp_rtcp_->GetNonSenderRttStats();
+  if (non_sender_rtt_stats.has_value()) {
+    stats.round_trip_time = non_sender_rtt_stats->round_trip_time;
+    stats.round_trip_time_measurements =
+        non_sender_rtt_stats->round_trip_time_measurements;
+    stats.total_round_trip_time = non_sender_rtt_stats->total_round_trip_time;
+  }
+
+  return stats;
+}
+
+void ChannelReceive::SetNACKStatus(bool enable, int max_packets) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // None of these functions can fail.
+  if (enable) {
+    rtp_receive_statistics_->SetMaxReorderingThreshold(max_packets);
+    acm_receiver_.EnableNack(max_packets);
+  } else {
+    rtp_receive_statistics_->SetMaxReorderingThreshold(
+        kDefaultMaxReorderingThreshold);
+    acm_receiver_.DisableNack();
+  }
+}
+
+void ChannelReceive::SetNonSenderRttMeasurement(bool enabled) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  rtp_rtcp_->SetNonSenderRttMeasurement(enabled);
+}
+
+// Called when we are missing one or more packets.
+int ChannelReceive::ResendPackets(const uint16_t* sequence_numbers,
+                                  int length) {
+  return rtp_rtcp_->SendNACK(sequence_numbers, length);
+}
+
+void ChannelReceive::RtcpPacketTypesCounterUpdated(
+    uint32_t ssrc,
+    const RtcpPacketTypeCounter& packet_counter) {
+  if (ssrc != remote_ssrc_) {
+    return;
+  }
+  MutexLock lock(&rtcp_counter_mutex_);
+  rtcp_packet_type_counter_ = packet_counter;
+}
+
+void ChannelReceive::SetAssociatedSendChannel(
+    const ChannelSendInterface* channel) {
+  RTC_DCHECK_RUN_ON(&network_thread_checker_);
+  associated_send_channel_ = channel;
+}
+
+void ChannelReceive::SetDepacketizerToDecoderFrameTransformer(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // Depending on when the channel is created, the transformer might be set
+  // twice. Don't replace the delegate if it was already initialized.
+  if (!frame_transformer || frame_transformer_delegate_) {
+    RTC_DCHECK_NOTREACHED() << "Not setting the transformer?";
+    return;
+  }
+
+  InitFrameTransformerDelegate(std::move(frame_transformer));
+}
+
+void ChannelReceive::SetFrameDecryptor(
+    rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
+  // TODO(bugs.webrtc.org/11993): Expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  frame_decryptor_ = std::move(frame_decryptor);
+}
+
+void ChannelReceive::OnLocalSsrcChange(uint32_t local_ssrc) {
+  // TODO(bugs.webrtc.org/11993): Expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  rtp_rtcp_->SetLocalSsrc(local_ssrc);
+}
+
+uint32_t ChannelReceive::GetLocalSsrc() const {
+  // TODO(bugs.webrtc.org/11993): Expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return rtp_rtcp_->local_media_ssrc();
+}
+
+NetworkStatistics ChannelReceive::GetNetworkStatistics(
+    bool get_and_clear_legacy_stats) const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  NetworkStatistics stats;
+  acm_receiver_.GetNetworkStatistics(&stats, get_and_clear_legacy_stats);
+  return stats;
+}
+
+AudioDecodingCallStats ChannelReceive::GetDecodingCallStatistics() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  AudioDecodingCallStats stats;
+  acm_receiver_.GetDecodingCallStatistics(&stats);
+  return stats;
+}
+
+uint32_t ChannelReceive::GetDelayEstimate() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // Return the current jitter buffer delay + playout delay.
+  return acm_receiver_.FilteredCurrentDelayMs() + playout_delay_ms_;
+}
+
+bool ChannelReceive::SetMinimumPlayoutDelay(int delay_ms) {
+  // TODO(bugs.webrtc.org/11993): This should run on the network thread.
+  // We get here via RtpStreamsSynchronizer. Once that's done, many (all?) of
+  // these locks aren't needed.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // Limit to range accepted by both VoE and ACM, so we're at least getting as
+  // close as possible, instead of failing.
+  delay_ms = rtc::SafeClamp(delay_ms, kVoiceEngineMinMinPlayoutDelayMs,
+                            kVoiceEngineMaxMinPlayoutDelayMs);
+  if (acm_receiver_.SetMinimumDelay(delay_ms) != 0) {
+    RTC_DLOG(LS_ERROR)
+        << "SetMinimumPlayoutDelay() failed to set min playout delay";
+    return false;
+  }
+  return true;
+}
+
+bool ChannelReceive::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+                                            int64_t* time_ms) const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!playout_timestamp_rtp_time_ms_)
+    return false;
+  *rtp_timestamp = playout_timestamp_rtp_;
+  *time_ms = playout_timestamp_rtp_time_ms_.value();
+  return true;
+}
+
+void ChannelReceive::SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
+                                                       int64_t time_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  playout_timestamp_ntp_ = ntp_timestamp_ms;
+  playout_timestamp_ntp_time_ms_ = time_ms;
+}
+
+absl::optional<int64_t>
+ChannelReceive::GetCurrentEstimatedPlayoutNtpTimestampMs(int64_t now_ms) const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!playout_timestamp_ntp_ || !playout_timestamp_ntp_time_ms_)
+    return absl::nullopt;
+
+  int64_t elapsed_ms = now_ms - *playout_timestamp_ntp_time_ms_;
+  return *playout_timestamp_ntp_ + elapsed_ms;
+}
+
+bool ChannelReceive::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
+  event_log_->Log(
+      std::make_unique<RtcEventNetEqSetMinimumDelay>(remote_ssrc_, delay_ms));
+  return acm_receiver_.SetBaseMinimumDelayMs(delay_ms);
+}
+
+int ChannelReceive::GetBaseMinimumPlayoutDelayMs() const {
+  return acm_receiver_.GetBaseMinimumDelayMs();
+}
+
+absl::optional<Syncable::Info> ChannelReceive::GetSyncInfo() const {
+  // TODO(bugs.webrtc.org/11993): This should run on the network thread.
+  // We get here via RtpStreamsSynchronizer. Once that's done, many of
+  // these locks aren't needed.
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  Syncable::Info info;
+  absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+      rtp_rtcp_->GetSenderReportStats();
+  if (!last_sr.has_value()) {
+    return absl::nullopt;
+  }
+  info.capture_time_ntp_secs = last_sr->last_remote_timestamp.seconds();
+  info.capture_time_ntp_frac = last_sr->last_remote_timestamp.fractions();
+  info.capture_time_source_clock = last_sr->last_remote_rtp_timestamp;
+
+  if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
+    return absl::nullopt;
+  }
+  info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
+  info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
+
+  int jitter_buffer_delay = acm_receiver_.FilteredCurrentDelayMs();
+  info.current_delay_ms = jitter_buffer_delay + playout_delay_ms_;
+
+  return info;
+}
+
+void ChannelReceive::UpdatePlayoutTimestamp(bool rtcp, int64_t now_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // TODO(bugs.webrtc.org/11993): Expect to be called exclusively on the
+  // network thread. Once that's done, we won't need video_sync_lock_.
+
+  jitter_buffer_playout_timestamp_ = acm_receiver_.GetPlayoutTimestamp();
+
+  if (!jitter_buffer_playout_timestamp_) {
+    // This can happen if this channel has not received any RTP packets. In
+    // this case, NetEq is not capable of computing a playout timestamp.
+    return;
+  }
+
+  uint16_t delay_ms = 0;
+  if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
+    RTC_DLOG(LS_WARNING)
+        << "ChannelReceive::UpdatePlayoutTimestamp() failed to read"
+           " playout delay from the ADM";
+    return;
+  }
+
+  RTC_DCHECK(jitter_buffer_playout_timestamp_);
+  uint32_t playout_timestamp = *jitter_buffer_playout_timestamp_;
+
+  // Remove the playout delay.
+  playout_timestamp -= (delay_ms * (GetRtpTimestampRateHz() / 1000));
+
+  if (!rtcp && playout_timestamp != playout_timestamp_rtp_) {
+    playout_timestamp_rtp_ = playout_timestamp;
+    playout_timestamp_rtp_time_ms_ = now_ms;
+  }
+  playout_delay_ms_ = delay_ms;
+}
+
+int ChannelReceive::GetRtpTimestampRateHz() const {
+  const auto decoder = acm_receiver_.LastDecoder();
+  // Default to the playout frequency if we've not gotten any packets yet.
+  // TODO(ossu): Zero clockrate can only happen if we've added an external
+  // decoder for a format we don't support internally. Remove once that way of
+  // adding decoders is gone!
+  // TODO(kwiberg): `decoder->second.clockrate_hz` is an RTP clockrate as it
+  // should, but `acm_receiver_.last_output_sample_rate_hz()` is a codec sample
+  // rate, which is not always the same thing.
+  return (decoder && decoder->second.clockrate_hz != 0)
+             ? decoder->second.clockrate_hz
+             : acm_receiver_.last_output_sample_rate_hz();
+}
+
+}  // namespace
+
+std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
+    Clock* clock,
+    NetEqFactory* neteq_factory,
+    AudioDeviceModule* audio_device_module,
+    Transport* rtcp_send_transport,
+    RtcEventLog* rtc_event_log,
+    uint32_t local_ssrc,
+    uint32_t remote_ssrc,
+    size_t jitter_buffer_max_packets,
+    bool jitter_buffer_fast_playout,
+    int jitter_buffer_min_delay_ms,
+    bool enable_non_sender_rtt,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+    absl::optional<AudioCodecPairId> codec_pair_id,
+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    RtcpEventObserver* rtcp_event_observer) {
+  return std::make_unique<ChannelReceive>(
+      clock, neteq_factory, audio_device_module, rtcp_send_transport,
+      rtc_event_log, local_ssrc, remote_ssrc, jitter_buffer_max_packets,
+      jitter_buffer_fast_playout, jitter_buffer_min_delay_ms,
+      enable_non_sender_rtt, decoder_factory, codec_pair_id,
+      std::move(frame_decryptor), crypto_options, std::move(frame_transformer),
+      rtcp_event_observer);
+}
+
+}  // namespace voe
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/channel_receive.h b/third_party/libwebrtc/audio/channel_receive.h
new file mode 100644
index 0000000000..5713d97aaa
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_receive.h
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_CHANNEL_RECEIVE_H_
+#define AUDIO_CHANNEL_RECEIVE_H_
+
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio/audio_mixer.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/call/audio_sink.h"
+#include "api/call/transport.h"
+#include "api/crypto/crypto_options.h"
+#include "api/frame_transformer_interface.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/transport/rtp/rtp_source.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "call/syncable.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/source_tracker.h"
+#include "system_wrappers/include/clock.h"
+
+// TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence
+// warnings about use of unsigned short.
+// These need cleanup, in a separate cl.
+
+namespace rtc {
+class TimestampWrapAroundHandler;
+}
+
+namespace webrtc {
+
+class AudioDeviceModule;
+class FrameDecryptorInterface;
+class PacketRouter;
+class RateLimiter;
+class ReceiveStatistics;
+class RtcEventLog;
+class RtpPacketReceived;
+class RtpRtcp;
+
+struct CallReceiveStatistics {
+  int cumulativeLost;
+  unsigned int jitterSamples;
+  int64_t payload_bytes_received = 0;
+  int64_t header_and_padding_bytes_received = 0;
+  int packetsReceived;
+  uint32_t nacks_sent = 0;
+  // The capture NTP time (in local timebase) of the first played out audio
+  // frame.
+  int64_t capture_start_ntp_time_ms_;
+  // The timestamp at which the last packet was received, i.e. the time of the
+  // local clock when it was received - not the RTP timestamp of that packet.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
+  absl::optional<Timestamp> last_packet_received;
+  // Remote outbound stats derived by the received RTCP sender reports.
+  // Note that the timestamps below correspond to the time elapsed since the
+  // Unix epoch.
+  // https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*
+  absl::optional<int64_t> last_sender_report_timestamp_ms;
+  absl::optional<int64_t> last_sender_report_remote_timestamp_ms;
+  uint64_t sender_reports_packets_sent = 0;
+  uint64_t sender_reports_bytes_sent = 0;
+  uint64_t sender_reports_reports_count = 0;
+  absl::optional<TimeDelta> round_trip_time;
+  TimeDelta total_round_trip_time = TimeDelta::Zero();
+  int round_trip_time_measurements;
+};
+
+namespace voe {
+
+class ChannelSendInterface;
+
+// Interface class needed for AudioReceiveStreamInterface tests that use a
+// MockChannelReceive.
+
+class ChannelReceiveInterface : public RtpPacketSinkInterface {
+ public:
+  virtual ~ChannelReceiveInterface() = default;
+
+  virtual void SetSink(AudioSinkInterface* sink) = 0;
+
+  virtual void SetReceiveCodecs(
+      const std::map<int, SdpAudioFormat>& codecs) = 0;
+
+  virtual void StartPlayout() = 0;
+  virtual void StopPlayout() = 0;
+
+  // Payload type and format of last received RTP packet, if any.
+  virtual absl::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec()
+      const = 0;
+
+  virtual void ReceivedRTCPPacket(const uint8_t* data, size_t length) = 0;
+
+  virtual void SetChannelOutputVolumeScaling(float scaling) = 0;
+  virtual int GetSpeechOutputLevelFullRange() const = 0;
+  // See description of "totalAudioEnergy" in the WebRTC stats spec:
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+  virtual double GetTotalOutputEnergy() const = 0;
+  virtual double GetTotalOutputDuration() const = 0;
+
+  // Stats.
+  virtual NetworkStatistics GetNetworkStatistics(
+      bool get_and_clear_legacy_stats) const = 0;
+  virtual AudioDecodingCallStats GetDecodingCallStatistics() const = 0;
+
+  // Audio+Video Sync.
+  virtual uint32_t GetDelayEstimate() const = 0;
+  virtual bool SetMinimumPlayoutDelay(int delay_ms) = 0;
+  virtual bool GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
+                                      int64_t* time_ms) const = 0;
+  virtual void SetEstimatedPlayoutNtpTimestampMs(int64_t ntp_timestamp_ms,
+                                                 int64_t time_ms) = 0;
+  virtual absl::optional<int64_t> GetCurrentEstimatedPlayoutNtpTimestampMs(
+      int64_t now_ms) const = 0;
+
+  // Audio quality.
+  // Base minimum delay sets lower bound on minimum delay value which
+  // determines minimum delay until audio playout.
+  virtual bool SetBaseMinimumPlayoutDelayMs(int delay_ms) = 0;
+  virtual int GetBaseMinimumPlayoutDelayMs() const = 0;
+
+  // Produces the transport-related timestamps; current_delay_ms is left unset.
+  virtual absl::optional<Syncable::Info> GetSyncInfo() const = 0;
+
+  virtual void RegisterReceiverCongestionControlObjects(
+      PacketRouter* packet_router) = 0;
+  virtual void ResetReceiverCongestionControlObjects() = 0;
+
+  virtual CallReceiveStatistics GetRTCPStatistics() const = 0;
+  virtual void SetNACKStatus(bool enable, int max_packets) = 0;
+  virtual void SetNonSenderRttMeasurement(bool enabled) = 0;
+
+  virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
+      int sample_rate_hz,
+      AudioFrame* audio_frame) = 0;
+
+  virtual int PreferredSampleRate() const = 0;
+
+  // Sets the source tracker to notify about "delivered" packets when output is
+  // muted.
+  virtual void SetSourceTracker(SourceTracker* source_tracker) = 0;
+
+  // Associate to a send channel.
+  // Used for obtaining RTT for a receive-only channel.
+  virtual void SetAssociatedSendChannel(
+      const ChannelSendInterface* channel) = 0;
+
+  // Sets a frame transformer between the depacketizer and the decoder, to
+  // transform the received frames before decoding them.
+  virtual void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+          frame_transformer) = 0;
+
+  virtual void SetFrameDecryptor(
+      rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
+
+  virtual void OnLocalSsrcChange(uint32_t local_ssrc) = 0;
+  virtual uint32_t GetLocalSsrc() const = 0;
+};
+
+std::unique_ptr<ChannelReceiveInterface> CreateChannelReceive(
+    Clock* clock,
+    NetEqFactory* neteq_factory,
+    AudioDeviceModule* audio_device_module,
+    Transport* rtcp_send_transport,
+    RtcEventLog* rtc_event_log,
+    uint32_t local_ssrc,
+    uint32_t remote_ssrc,
+    size_t jitter_buffer_max_packets,
+    bool jitter_buffer_fast_playout,
+    int jitter_buffer_min_delay_ms,
+    bool enable_non_sender_rtt,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+    absl::optional<AudioCodecPairId> codec_pair_id,
+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    RtcpEventObserver* rtcp_event_observer);
+
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // AUDIO_CHANNEL_RECEIVE_H_
diff --git a/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.cc b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.cc
new file mode 100644
index 0000000000..2d2893b8f7
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/channel_receive_frame_transformer_delegate.h"
+
+#include <utility>
+
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace {
+
+class TransformableIncomingAudioFrame
+    : public TransformableAudioFrameInterface {
+ public:
+  TransformableIncomingAudioFrame(rtc::ArrayView<const uint8_t> payload,
+                                  const RTPHeader& header,
+                                  uint32_t ssrc)
+      : payload_(payload.data(), payload.size()),
+        header_(header),
+        ssrc_(ssrc) {}
+  ~TransformableIncomingAudioFrame() override = default;
+  rtc::ArrayView<const uint8_t> GetData() const override { return payload_; }
+
+  void SetData(rtc::ArrayView<const uint8_t> data) override {
+    payload_.SetData(data.data(), data.size());
+  }
+
+  void SetRTPTimestamp(uint32_t timestamp) override {
+    header_.timestamp = timestamp;
+  }
+
+  uint8_t GetPayloadType() const override { return header_.payloadType; }
+  uint32_t GetSsrc() const override { return ssrc_; }
+  uint32_t GetTimestamp() const override { return header_.timestamp; }
+  rtc::ArrayView<const uint32_t> GetContributingSources() const override {
+    return rtc::ArrayView<const uint32_t>(header_.arrOfCSRCs, header_.numCSRCs);
+  }
+  Direction GetDirection() const override { return Direction::kReceiver; }
+
+  const absl::optional<uint16_t> SequenceNumber() const override {
+    return header_.sequenceNumber;
+  }
+
+  absl::optional<uint64_t> AbsoluteCaptureTimestamp() const override {
+    // This could be extracted from received header extensions + extrapolation,
+    // if required in future, eg for being able to re-send received frames.
+    return absl::nullopt;
+  }
+  const RTPHeader& Header() const { return header_; }
+
+  FrameType Type() const override {
+    return header_.extension.voiceActivity ? FrameType::kAudioFrameSpeech
+                                           : FrameType::kAudioFrameCN;
+  }
+
+ private:
+  rtc::Buffer payload_;
+  RTPHeader header_;
+  uint32_t ssrc_;
+};
+}  // namespace
+
+ChannelReceiveFrameTransformerDelegate::ChannelReceiveFrameTransformerDelegate(
+    ReceiveFrameCallback receive_frame_callback,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    TaskQueueBase* channel_receive_thread)
+    : receive_frame_callback_(receive_frame_callback),
+      frame_transformer_(std::move(frame_transformer)),
+      channel_receive_thread_(channel_receive_thread) {}
+
+void ChannelReceiveFrameTransformerDelegate::Init() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  frame_transformer_->RegisterTransformedFrameCallback(
+      rtc::scoped_refptr<TransformedFrameCallback>(this));
+}
+
+void ChannelReceiveFrameTransformerDelegate::Reset() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  frame_transformer_->UnregisterTransformedFrameCallback();
+  frame_transformer_ = nullptr;
+  receive_frame_callback_ = ReceiveFrameCallback();
+}
+
+void ChannelReceiveFrameTransformerDelegate::Transform(
+    rtc::ArrayView<const uint8_t> packet,
+    const RTPHeader& header,
+    uint32_t ssrc) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  frame_transformer_->Transform(
+      std::make_unique<TransformableIncomingAudioFrame>(packet, header, ssrc));
+}
+
+void ChannelReceiveFrameTransformerDelegate::OnTransformedFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) {
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate(this);
+  channel_receive_thread_->PostTask(
+      [delegate = std::move(delegate), frame = std::move(frame)]() mutable {
+        delegate->ReceiveFrame(std::move(frame));
+      });
+}
+
+void ChannelReceiveFrameTransformerDelegate::ReceiveFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  if (!receive_frame_callback_)
+    return;
+
+  RTPHeader header;
+  if (frame->GetDirection() ==
+      TransformableFrameInterface::Direction::kSender) {
+    auto* transformed_frame =
+        static_cast<TransformableAudioFrameInterface*>(frame.get());
+    header.payloadType = transformed_frame->GetPayloadType();
+    header.timestamp = transformed_frame->GetTimestamp();
+    header.ssrc = transformed_frame->GetSsrc();
+    if (transformed_frame->AbsoluteCaptureTimestamp().has_value()) {
+      header.extension.absolute_capture_time = AbsoluteCaptureTime();
+      header.extension.absolute_capture_time->absolute_capture_timestamp =
+          transformed_frame->AbsoluteCaptureTimestamp().value();
+    }
+  } else {
+    auto* transformed_frame =
+        static_cast<TransformableIncomingAudioFrame*>(frame.get());
+    header = transformed_frame->Header();
+  }
+
+  // TODO(crbug.com/1464860): Take an explicit struct with the required
+  // information rather than the RTPHeader to make it easier to
+  // construct the required information when injecting transformed frames not
+  // originally from this receiver.
+  receive_frame_callback_(frame->GetData(), header);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.h b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.h
new file mode 100644
index 0000000000..04ad7c4695
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_CHANNEL_RECEIVE_FRAME_TRANSFORMER_DELEGATE_H_
+#define AUDIO_CHANNEL_RECEIVE_FRAME_TRANSFORMER_DELEGATE_H_
+
+#include <memory>
+
+#include "api/frame_transformer_interface.h"
+#include "api/sequence_checker.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+// Delegates calls to FrameTransformerInterface to transform frames, and to
+// ChannelReceive to receive the transformed frames using the
+// `receive_frame_callback_` on the `channel_receive_thread_`.
+class ChannelReceiveFrameTransformerDelegate : public TransformedFrameCallback {
+ public:
+  using ReceiveFrameCallback =
+      std::function<void(rtc::ArrayView<const uint8_t> packet,
+                         const RTPHeader& header)>;
+  ChannelReceiveFrameTransformerDelegate(
+      ReceiveFrameCallback receive_frame_callback,
+      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+      TaskQueueBase* channel_receive_thread);
+
+  // Registers `this` as callback for `frame_transformer_`, to get the
+  // transformed frames.
+  void Init();
+
+  // Unregisters and releases the `frame_transformer_` reference, and resets
+  // `receive_frame_callback_` on `channel_receive_thread_`. Called from
+  // ChannelReceive destructor to prevent running the callback on a dangling
+  // channel.
+  void Reset();
+
+  // Delegates the call to FrameTransformerInterface::Transform, to transform
+  // the frame asynchronously.
+  void Transform(rtc::ArrayView<const uint8_t> packet,
+                 const RTPHeader& header,
+                 uint32_t ssrc);
+
+  // Implements TransformedFrameCallback. Can be called on any thread.
+  void OnTransformedFrame(
+      std::unique_ptr<TransformableFrameInterface> frame) override;
+
+  // Delegates the call to ChannelReceive::OnReceivedPayloadData on the
+  // `channel_receive_thread_`, by calling `receive_frame_callback_`.
+  void ReceiveFrame(std::unique_ptr<TransformableFrameInterface> frame) const;
+
+ protected:
+  ~ChannelReceiveFrameTransformerDelegate() override = default;
+
+ private:
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+  ReceiveFrameCallback receive_frame_callback_
+      RTC_GUARDED_BY(sequence_checker_);
+  rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
+      RTC_GUARDED_BY(sequence_checker_);
+  TaskQueueBase* const channel_receive_thread_;
+};
+
+}  // namespace webrtc
+#endif  // AUDIO_CHANNEL_RECEIVE_FRAME_TRANSFORMER_DELEGATE_H_
diff --git a/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate_unittest.cc b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate_unittest.cc
new file mode 100644
index 0000000000..38ceb6d96d
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_receive_frame_transformer_delegate_unittest.cc
@@ -0,0 +1,154 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/channel_receive_frame_transformer_delegate.h"
+
+#include <memory>
+#include <utility>
+
+#include "audio/channel_send_frame_transformer_delegate.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_frame_transformer.h"
+#include "test/mock_transformable_frame.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::NiceMock;
+using ::testing::SaveArg;
+
+class MockChannelReceive {
+ public:
+  MOCK_METHOD(void,
+              ReceiveFrame,
+              (rtc::ArrayView<const uint8_t> packet, const RTPHeader& header));
+
+  ChannelReceiveFrameTransformerDelegate::ReceiveFrameCallback callback() {
+    return [this](rtc::ArrayView<const uint8_t> packet,
+                  const RTPHeader& header) { ReceiveFrame(packet, header); };
+  }
+};
+
+// Test that the delegate registers itself with the frame transformer on Init().
+TEST(ChannelReceiveFrameTransformerDelegateTest,
+     RegisterTransformedFrameCallbackOnInit) {
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<MockFrameTransformer>();
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
+          ChannelReceiveFrameTransformerDelegate::ReceiveFrameCallback(),
+          mock_frame_transformer, nullptr);
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback);
+  delegate->Init();
+}
+
+// Test that the delegate unregisters itself from the frame transformer on
+// Reset().
+TEST(ChannelReceiveFrameTransformerDelegateTest,
+     UnregisterTransformedFrameCallbackOnReset) {
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<MockFrameTransformer>();
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
+          ChannelReceiveFrameTransformerDelegate::ReceiveFrameCallback(),
+          mock_frame_transformer, nullptr);
+  EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
+  delegate->Reset();
+}
+
+// Test that when the delegate receives a transformed frame from the frame
+// transformer, it passes it to the channel using the ReceiveFrameCallback.
+TEST(ChannelReceiveFrameTransformerDelegateTest,
+     TransformRunsChannelReceiveCallback) {
+  rtc::AutoThread main_thread;
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  MockChannelReceive mock_channel;
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
+          mock_channel.callback(), mock_frame_transformer,
+          rtc::Thread::Current());
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  delegate->Init();
+  ASSERT_TRUE(callback);
+
+  const uint8_t data[] = {1, 2, 3, 4};
+  rtc::ArrayView<const uint8_t> packet(data, sizeof(data));
+  RTPHeader header;
+  EXPECT_CALL(mock_channel, ReceiveFrame);
+  ON_CALL(*mock_frame_transformer, Transform)
+      .WillByDefault(
+          [&callback](std::unique_ptr<TransformableFrameInterface> frame) {
+            callback->OnTransformedFrame(std::move(frame));
+          });
+  delegate->Transform(packet, header, 1111 /*ssrc*/);
+  rtc::ThreadManager::ProcessAllMessageQueuesForTesting();
+}
+
+// Test that when the delegate receives a Outgoing frame from the frame
+// transformer, it passes it to the channel using the ReceiveFrameCallback.
+TEST(ChannelReceiveFrameTransformerDelegateTest,
+     TransformRunsChannelReceiveCallbackForSenderFrame) {
+  rtc::AutoThread main_thread;
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  MockChannelReceive mock_channel;
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
+          mock_channel.callback(), mock_frame_transformer,
+          rtc::Thread::Current());
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  delegate->Init();
+  ASSERT_TRUE(callback);
+
+  const uint8_t data[] = {1, 2, 3, 4};
+  rtc::ArrayView<const uint8_t> packet(data, sizeof(data));
+  RTPHeader header;
+  EXPECT_CALL(mock_channel, ReceiveFrame(ElementsAre(1, 2, 3, 4), _));
+  ON_CALL(*mock_frame_transformer, Transform)
+      .WillByDefault([&callback](
+                         std::unique_ptr<TransformableFrameInterface> frame) {
+        auto* transformed_frame =
+            static_cast<TransformableAudioFrameInterface*>(frame.get());
+        callback->OnTransformedFrame(CloneSenderAudioFrame(transformed_frame));
+      });
+  delegate->Transform(packet, header, 1111 /*ssrc*/);
+  rtc::ThreadManager::ProcessAllMessageQueuesForTesting();
+}
+
+// Test that if the delegate receives a transformed frame after it has been
+// reset, it does not run the ReceiveFrameCallback, as the channel is destroyed
+// after resetting the delegate.
+TEST(ChannelReceiveFrameTransformerDelegateTest,
+     OnTransformedDoesNotRunChannelReceiveCallbackAfterReset) {
+  rtc::AutoThread main_thread;
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<testing::NiceMock<MockFrameTransformer>>();
+  MockChannelReceive mock_channel;
+  rtc::scoped_refptr<ChannelReceiveFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelReceiveFrameTransformerDelegate>(
+          mock_channel.callback(), mock_frame_transformer,
+          rtc::Thread::Current());
+
+  delegate->Reset();
+  EXPECT_CALL(mock_channel, ReceiveFrame).Times(0);
+  delegate->OnTransformedFrame(std::make_unique<MockTransformableAudioFrame>());
+  rtc::ThreadManager::ProcessAllMessageQueuesForTesting();
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/channel_receive_unittest.cc b/third_party/libwebrtc/audio/channel_receive_unittest.cc
new file mode 100644
index 0000000000..4b7b7c0231
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_receive_unittest.cc
@@ -0,0 +1,231 @@
+/*
+ *  Copyright 2023 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/channel_receive.h"
+
+#include "absl/strings/escaping.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/thread.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder_factory.h"
+#include "test/mock_transport.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace voe {
+namespace {
+
+using ::testing::NiceMock;
+using ::testing::NotNull;
+using ::testing::Return;
+using ::testing::Test;
+
+constexpr uint32_t kLocalSsrc = 1111;
+constexpr uint32_t kRemoteSsrc = 2222;
+// We run RTP data with 8 kHz PCMA (fixed payload type 8).
+constexpr char kPayloadName[] = "PCMA";
+constexpr int kPayloadType = 8;
+constexpr int kSampleRateHz = 8000;
+
+class ChannelReceiveTest : public Test {
+ public:
+  ChannelReceiveTest()
+      : time_controller_(Timestamp::Seconds(5555)),
+        audio_device_module_(test::MockAudioDeviceModule::CreateNice()),
+        audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()) {
+    ON_CALL(*audio_device_module_, PlayoutDelay).WillByDefault(Return(0));
+  }
+
+  std::unique_ptr<ChannelReceiveInterface> CreateTestChannelReceive() {
+    CryptoOptions crypto_options;
+    auto channel = CreateChannelReceive(
+        time_controller_.GetClock(),
+        /* neteq_factory= */ nullptr, audio_device_module_.get(), &transport_,
+        &event_log_, kLocalSsrc, kRemoteSsrc,
+        /* jitter_buffer_max_packets= */ 0,
+        /* jitter_buffer_fast_playout= */ false,
+        /* jitter_buffer_min_delay_ms= */ 0,
+        /* enable_non_sender_rtt= */ false, audio_decoder_factory_,
+        /* codec_pair_id= */ absl::nullopt,
+        /* frame_decryptor_interface= */ nullptr, crypto_options,
+        /* frame_transformer= */ nullptr);
+    channel->SetReceiveCodecs(
+        {{kPayloadType, {kPayloadName, kSampleRateHz, 1}}});
+    return channel;
+  }
+
+  NtpTime NtpNow() { return time_controller_.GetClock()->CurrentNtpTime(); }
+
+  uint32_t RtpNow() {
+    // Note - the "random" offset of this timestamp is zero.
+    return rtc::TimeMillis() * 1000 / kSampleRateHz;
+  }
+
+  RtpPacketReceived CreateRtpPacket() {
+    RtpPacketReceived packet;
+    packet.set_arrival_time(time_controller_.GetClock()->CurrentTime());
+    packet.SetTimestamp(RtpNow());
+    packet.SetSsrc(kLocalSsrc);
+    packet.SetPayloadType(kPayloadType);
+    // Packet size should be enough to give at least 10 ms of data.
+    // For PCMA, that's 80 bytes; this should be enough.
+    uint8_t* datapos = packet.SetPayloadSize(100);
+    memset(datapos, 0, 100);
+    return packet;
+  }
+
+  std::vector<uint8_t> CreateRtcpSenderReport() {
+    std::vector<uint8_t> packet(1024);
+    size_t pos = 0;
+    rtcp::SenderReport report;
+    report.SetSenderSsrc(kRemoteSsrc);
+    report.SetNtp(NtpNow());
+    report.SetRtpTimestamp(RtpNow());
+    report.SetPacketCount(0);
+    report.SetOctetCount(0);
+    report.Create(&packet[0], &pos, packet.size(), nullptr);
+    // No report blocks.
+    packet.resize(pos);
+    return packet;
+  }
+
+  std::vector<uint8_t> CreateRtcpReceiverReport() {
+    rtcp::ReportBlock block;
+    block.SetMediaSsrc(kLocalSsrc);
+    // Middle 32 bits of the NTP timestamp from received SR
+    block.SetLastSr(CompactNtp(NtpNow()));
+    block.SetDelayLastSr(0);
+
+    rtcp::ReceiverReport report;
+    report.SetSenderSsrc(kRemoteSsrc);
+    report.AddReportBlock(block);
+
+    std::vector<uint8_t> packet(1024);
+    size_t pos = 0;
+    report.Create(&packet[0], &pos, packet.size(), nullptr);
+    packet.resize(pos);
+    return packet;
+  }
+
+  void HandleGeneratedRtcp(ChannelReceiveInterface& channel,
+                           rtc::ArrayView<const uint8_t> packet) {
+    if (packet[1] == rtcp::ReceiverReport::kPacketType) {
+      // Ignore RR, it requires no response
+    } else {
+      RTC_LOG(LS_ERROR) << "Unexpected RTCP packet generated";
+      RTC_LOG(LS_ERROR) << "Packet content "
+                        << rtc::hex_encode_with_delimiter(
+                               absl::string_view(
+                                   reinterpret_cast<char*>(packet.data()[0]),
+                                   packet.size()),
+                               ' ');
+    }
+  }
+
+  int64_t ProbeCaptureStartNtpTime(ChannelReceiveInterface& channel) {
+    // Computation of the capture_start_ntp_time_ms_ occurs when the
+    // audio data is pulled, not when it is received. So we need to
+    // inject an RTP packet, and then fetch its data.
+    AudioFrame audio_frame;
+    channel.OnRtpPacket(CreateRtpPacket());
+    channel.GetAudioFrameWithInfo(kSampleRateHz, &audio_frame);
+    CallReceiveStatistics stats = channel.GetRTCPStatistics();
+    return stats.capture_start_ntp_time_ms_;
+  }
+
+ protected:
+  GlobalSimulatedTimeController time_controller_;
+  rtc::scoped_refptr<test::MockAudioDeviceModule> audio_device_module_;
+  rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
+  MockTransport transport_;
+  NiceMock<MockRtcEventLog> event_log_;
+};
+
+TEST_F(ChannelReceiveTest, CreateAndDestroy) {
+  auto channel = CreateTestChannelReceive();
+  EXPECT_THAT(channel, NotNull());
+}
+
+TEST_F(ChannelReceiveTest, ReceiveReportGeneratedOnTime) {
+  auto channel = CreateTestChannelReceive();
+
+  bool receiver_report_sent = false;
+  EXPECT_CALL(transport_, SendRtcp)
+      .WillRepeatedly([&](rtc::ArrayView<const uint8_t> packet) {
+        if (packet.size() >= 2 &&
+            packet[1] == rtcp::ReceiverReport::kPacketType) {
+          receiver_report_sent = true;
+        }
+        return true;
+      });
+  // RFC 3550 section 6.2 mentions 5 seconds as a reasonable expectation
+  // for the interval between RTCP packets.
+  time_controller_.AdvanceTime(TimeDelta::Seconds(5));
+
+  EXPECT_TRUE(receiver_report_sent);
+}
+
+TEST_F(ChannelReceiveTest, CaptureStartTimeBecomesValid) {
+  auto channel = CreateTestChannelReceive();
+
+  EXPECT_CALL(transport_, SendRtcp)
+      .WillRepeatedly([&](rtc::ArrayView<const uint8_t> packet) {
+        HandleGeneratedRtcp(*channel, packet);
+        return true;
+      });
+  // Before any packets are sent, CaptureStartTime is invalid.
+  EXPECT_EQ(ProbeCaptureStartNtpTime(*channel), -1);
+
+  // Must start playout, otherwise packet is discarded.
+  channel->StartPlayout();
+  // Send one RTP packet. This causes registration of the SSRC.
+  channel->OnRtpPacket(CreateRtpPacket());
+  EXPECT_EQ(ProbeCaptureStartNtpTime(*channel), -1);
+
+  // Receive a sender report.
+  auto rtcp_packet_1 = CreateRtcpSenderReport();
+  channel->ReceivedRTCPPacket(rtcp_packet_1.data(), rtcp_packet_1.size());
+  EXPECT_EQ(ProbeCaptureStartNtpTime(*channel), -1);
+
+  time_controller_.AdvanceTime(TimeDelta::Seconds(5));
+
+  // Receive a receiver report. This is necessary, which is odd.
+  // Presumably it is because the receiver needs to know the RTT
+  // before it can compute the capture start NTP time.
+  // The receiver report must happen before the second sender report.
+  auto rtcp_rr = CreateRtcpReceiverReport();
+  channel->ReceivedRTCPPacket(rtcp_rr.data(), rtcp_rr.size());
+  EXPECT_EQ(ProbeCaptureStartNtpTime(*channel), -1);
+
+  // Receive another sender report after 5 seconds.
+  // This should be enough to establish the capture start NTP time.
+  auto rtcp_packet_2 = CreateRtcpSenderReport();
+  channel->ReceivedRTCPPacket(rtcp_packet_2.data(), rtcp_packet_2.size());
+
+  EXPECT_NE(ProbeCaptureStartNtpTime(*channel), -1);
+}
+
+}  // namespace
+}  // namespace voe
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/channel_send.cc b/third_party/libwebrtc/audio/channel_send.cc
new file mode 100644
index 0000000000..ee94760b6f
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_send.cc
@@ -0,0 +1,922 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/channel_send.h"
+
+#include <algorithm>
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/call/transport.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/sequence_checker.h"
+#include "audio/channel_send_frame_transformer_delegate.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_processing/rms_level.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/rate_limiter.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace voe {
+
+namespace {
+
+constexpr int64_t kMaxRetransmissionWindowMs = 1000;
+constexpr int64_t kMinRetransmissionWindowMs = 30;
+
+class RtpPacketSenderProxy;
+class TransportSequenceNumberProxy;
+
+class RtcpCounterObserver : public RtcpPacketTypeCounterObserver {
+ public:
+  explicit RtcpCounterObserver(uint32_t ssrc) : ssrc_(ssrc) {}
+
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc, const RtcpPacketTypeCounter& packet_counter) override {
+    if (ssrc_ != ssrc) {
+      return;
+    }
+
+    MutexLock lock(&mutex_);
+    packet_counter_ = packet_counter;
+  }
+
+  RtcpPacketTypeCounter GetCounts() {
+    MutexLock lock(&mutex_);
+    return packet_counter_;
+  }
+
+ private:
+  Mutex mutex_;
+  const uint32_t ssrc_;
+  RtcpPacketTypeCounter packet_counter_;
+};
+
+class ChannelSend : public ChannelSendInterface,
+                    public AudioPacketizationCallback,  // receive encoded
+                                                        // packets from the ACM
+                    public RtcpPacketTypeCounterObserver,
+                    public ReportBlockDataObserver {
+ public:
+  ChannelSend(Clock* clock,
+              TaskQueueFactory* task_queue_factory,
+              Transport* rtp_transport,
+              RtcpRttStats* rtcp_rtt_stats,
+              RtcEventLog* rtc_event_log,
+              FrameEncryptorInterface* frame_encryptor,
+              const webrtc::CryptoOptions& crypto_options,
+              bool extmap_allow_mixed,
+              int rtcp_report_interval_ms,
+              uint32_t ssrc,
+              rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+              RtpTransportControllerSendInterface* transport_controller,
+              const FieldTrialsView& field_trials);
+
+  ~ChannelSend() override;
+
+  // Send using this encoder, with this payload type.
+  void SetEncoder(int payload_type,
+                  std::unique_ptr<AudioEncoder> encoder) override;
+  void ModifyEncoder(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)>
+                         modifier) override;
+  void CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) override;
+
+  // API methods
+  void StartSend() override;
+  void StopSend() override;
+
+  // Codecs
+  void OnBitrateAllocation(BitrateAllocationUpdate update) override;
+  int GetTargetBitrate() const override;
+
+  // Network
+  void ReceivedRTCPPacket(const uint8_t* data, size_t length) override;
+
+  // Muting, Volume and Level.
+  void SetInputMute(bool enable) override;
+
+  // Stats.
+  ANAStats GetANAStatistics() const override;
+
+  // Used by AudioSendStream.
+  RtpRtcpInterface* GetRtpRtcp() const override;
+
+  void RegisterCngPayloadType(int payload_type, int payload_frequency) override;
+
+  // DTMF.
+  bool SendTelephoneEventOutband(int event, int duration_ms) override;
+  void SetSendTelephoneEventPayloadType(int payload_type,
+                                        int payload_frequency) override;
+
+  // RTP+RTCP
+  void SetSendAudioLevelIndicationStatus(bool enable, int id) override;
+
+  void RegisterSenderCongestionControlObjects(
+      RtpTransportControllerSendInterface* transport) override;
+  void ResetSenderCongestionControlObjects() override;
+  void SetRTCP_CNAME(absl::string_view c_name) override;
+  std::vector<ReportBlockData> GetRemoteRTCPReportBlocks() const override;
+  CallSendStatistics GetRTCPStatistics() const override;
+
+  // ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
+  // which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
+  // the actual processing of the audio takes place. The processing mainly
+  // consists of encoding and preparing the result for sending by adding it to a
+  // send queue.
+  // The main reason for using a task queue here is to release the native,
+  // OS-specific, audio capture thread as soon as possible to ensure that it
+  // can go back to sleep and be prepared to deliver an new captured audio
+  // packet.
+  void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) override;
+
+  int64_t GetRTT() const override;
+
+  // E2EE Custom Audio Frame Encryption
+  void SetFrameEncryptor(
+      rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) override;
+
+  // Sets a frame transformer between encoder and packetizer, to transform
+  // encoded frames before sending them out the network.
+  void SetEncoderToPacketizerFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer)
+      override;
+
+  // RtcpPacketTypeCounterObserver.
+  void RtcpPacketTypesCounterUpdated(
+      uint32_t ssrc,
+      const RtcpPacketTypeCounter& packet_counter) override;
+
+  // ReportBlockDataObserver.
+  void OnReportBlockDataUpdated(ReportBlockData report_block) override;
+
+ private:
+  // From AudioPacketizationCallback in the ACM
+  int32_t SendData(AudioFrameType frameType,
+                   uint8_t payloadType,
+                   uint32_t rtp_timestamp,
+                   const uint8_t* payloadData,
+                   size_t payloadSize,
+                   int64_t absolute_capture_timestamp_ms) override;
+
+  bool InputMute() const;
+
+  int32_t SendRtpAudio(AudioFrameType frameType,
+                       uint8_t payloadType,
+                       uint32_t rtp_timestamp_without_offset,
+                       rtc::ArrayView<const uint8_t> payload,
+                       int64_t absolute_capture_timestamp_ms)
+      RTC_RUN_ON(encoder_queue_);
+
+  void OnReceivedRtt(int64_t rtt_ms);
+
+  void InitFrameTransformerDelegate(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer);
+
+  // Thread checkers document and lock usage of some methods on voe::Channel to
+  // specific threads we know about. The goal is to eventually split up
+  // voe::Channel into parts with single-threaded semantics, and thereby reduce
+  // the need for locks.
+  SequenceChecker worker_thread_checker_;
+  // Methods accessed from audio and video threads are checked for sequential-
+  // only access. We don't necessarily own and control these threads, so thread
+  // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
+  // audio thread to another, but access is still sequential.
+  rtc::RaceChecker audio_thread_race_checker_;
+
+  mutable Mutex volume_settings_mutex_;
+
+  const uint32_t ssrc_;
+  bool sending_ RTC_GUARDED_BY(&worker_thread_checker_) = false;
+
+  RtcEventLog* const event_log_;
+
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
+  std::unique_ptr<RTPSenderAudio> rtp_sender_audio_;
+
+  std::unique_ptr<AudioCodingModule> audio_coding_;
+
+  // This is just an offset, RTP module will add its own random offset.
+  uint32_t timestamp_ RTC_GUARDED_BY(audio_thread_race_checker_) = 0;
+  absl::optional<int64_t> last_capture_timestamp_ms_
+      RTC_GUARDED_BY(audio_thread_race_checker_);
+
+  RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
+  bool input_mute_ RTC_GUARDED_BY(volume_settings_mutex_) = false;
+  bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_) = false;
+
+  const std::unique_ptr<RtcpCounterObserver> rtcp_counter_observer_;
+
+  PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
+      nullptr;
+  const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_pacer_proxy_;
+  const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
+
+  SequenceChecker construction_thread_;
+
+  std::atomic<bool> include_audio_level_indication_ = false;
+  std::atomic<bool> encoder_queue_is_active_ = false;
+  std::atomic<bool> first_frame_ = true;
+
+  // E2EE Audio Frame Encryption
+  rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_
+      RTC_GUARDED_BY(encoder_queue_);
+  // E2EE Frame Encryption Options
+  const webrtc::CryptoOptions crypto_options_;
+
+  // Delegates calls to a frame transformer to transform audio, and
+  // receives callbacks with the transformed frames; delegates calls to
+  // ChannelSend::SendRtpAudio to send the transformed audio.
+  rtc::scoped_refptr<ChannelSendFrameTransformerDelegate>
+      frame_transformer_delegate_ RTC_GUARDED_BY(encoder_queue_);
+
+  mutable Mutex rtcp_counter_mutex_;
+  RtcpPacketTypeCounter rtcp_packet_type_counter_
+      RTC_GUARDED_BY(rtcp_counter_mutex_);
+
+  // Defined last to ensure that there are no running tasks when the other
+  // members are destroyed.
+  rtc::TaskQueue encoder_queue_;
+};
+
+const int kTelephoneEventAttenuationdB = 10;
+
+class RtpPacketSenderProxy : public RtpPacketSender {
+ public:
+  RtpPacketSenderProxy() : rtp_packet_pacer_(nullptr) {}
+
+  void SetPacketPacer(RtpPacketSender* rtp_packet_pacer) {
+    RTC_DCHECK(thread_checker_.IsCurrent());
+    MutexLock lock(&mutex_);
+    rtp_packet_pacer_ = rtp_packet_pacer;
+  }
+
+  void EnqueuePackets(
+      std::vector<std::unique_ptr<RtpPacketToSend>> packets) override {
+    MutexLock lock(&mutex_);
+    if (rtp_packet_pacer_) {
+      rtp_packet_pacer_->EnqueuePackets(std::move(packets));
+    }
+  }
+
+  void RemovePacketsForSsrc(uint32_t ssrc) override {
+    MutexLock lock(&mutex_);
+    if (rtp_packet_pacer_) {
+      rtp_packet_pacer_->RemovePacketsForSsrc(ssrc);
+    }
+  }
+
+ private:
+  SequenceChecker thread_checker_;
+  Mutex mutex_;
+  RtpPacketSender* rtp_packet_pacer_ RTC_GUARDED_BY(&mutex_);
+};
+
+int32_t ChannelSend::SendData(AudioFrameType frameType,
+                              uint8_t payloadType,
+                              uint32_t rtp_timestamp,
+                              const uint8_t* payloadData,
+                              size_t payloadSize,
+                              int64_t absolute_capture_timestamp_ms) {
+  RTC_DCHECK_RUN_ON(&encoder_queue_);
+  rtc::ArrayView<const uint8_t> payload(payloadData, payloadSize);
+  if (frame_transformer_delegate_) {
+    // Asynchronously transform the payload before sending it. After the payload
+    // is transformed, the delegate will call SendRtpAudio to send it.
+    frame_transformer_delegate_->Transform(
+        frameType, payloadType, rtp_timestamp + rtp_rtcp_->StartTimestamp(),
+        payloadData, payloadSize, absolute_capture_timestamp_ms,
+        rtp_rtcp_->SSRC());
+    return 0;
+  }
+  return SendRtpAudio(frameType, payloadType, rtp_timestamp, payload,
+                      absolute_capture_timestamp_ms);
+}
+
+int32_t ChannelSend::SendRtpAudio(AudioFrameType frameType,
+                                  uint8_t payloadType,
+                                  uint32_t rtp_timestamp_without_offset,
+                                  rtc::ArrayView<const uint8_t> payload,
+                                  int64_t absolute_capture_timestamp_ms) {
+  // E2EE Custom Audio Frame Encryption (This is optional).
+  // Keep this buffer around for the lifetime of the send call.
+  rtc::Buffer encrypted_audio_payload;
+  // We don't invoke encryptor if payload is empty, which means we are to send
+  // DTMF, or the encoder entered DTX.
+  // TODO(minyue): see whether DTMF packets should be encrypted or not. In
+  // current implementation, they are not.
+  if (!payload.empty()) {
+    if (frame_encryptor_ != nullptr) {
+      // TODO(benwright@webrtc.org) - Allocate enough to always encrypt inline.
+      // Allocate a buffer to hold the maximum possible encrypted payload.
+      size_t max_ciphertext_size = frame_encryptor_->GetMaxCiphertextByteSize(
+          cricket::MEDIA_TYPE_AUDIO, payload.size());
+      encrypted_audio_payload.SetSize(max_ciphertext_size);
+
+      // Encrypt the audio payload into the buffer.
+      size_t bytes_written = 0;
+      int encrypt_status = frame_encryptor_->Encrypt(
+          cricket::MEDIA_TYPE_AUDIO, rtp_rtcp_->SSRC(),
+          /*additional_data=*/nullptr, payload, encrypted_audio_payload,
+          &bytes_written);
+      if (encrypt_status != 0) {
+        RTC_DLOG(LS_ERROR)
+            << "Channel::SendData() failed encrypt audio payload: "
+            << encrypt_status;
+        return -1;
+      }
+      // Resize the buffer to the exact number of bytes actually used.
+      encrypted_audio_payload.SetSize(bytes_written);
+      // Rewrite the payloadData and size to the new encrypted payload.
+      payload = encrypted_audio_payload;
+    } else if (crypto_options_.sframe.require_frame_encryption) {
+      RTC_DLOG(LS_ERROR) << "Channel::SendData() failed sending audio payload: "
+                            "A frame encryptor is required but one is not set.";
+      return -1;
+    }
+  }
+
+  // Push data from ACM to RTP/RTCP-module to deliver audio frame for
+  // packetization.
+  if (!rtp_rtcp_->OnSendingRtpFrame(rtp_timestamp_without_offset,
+                                    // Leaving the time when this frame was
+                                    // received from the capture device as
+                                    // undefined for voice for now.
+                                    -1, payloadType,
+                                    /*force_sender_report=*/false)) {
+    return -1;
+  }
+
+  // RTCPSender has it's own copy of the timestamp offset, added in
+  // RTCPSender::BuildSR, hence we must not add the in the offset for the above
+  // call.
+  // TODO(nisse): Delete RTCPSender:timestamp_offset_, and see if we can confine
+  // knowledge of the offset to a single place.
+
+  // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
+  RTPSenderAudio::RtpAudioFrame frame = {
+      .type = frameType,
+      .payload = payload,
+      .payload_id = payloadType,
+      .rtp_timestamp =
+          rtp_timestamp_without_offset + rtp_rtcp_->StartTimestamp()};
+  if (absolute_capture_timestamp_ms > 0) {
+    frame.capture_time = Timestamp::Millis(absolute_capture_timestamp_ms);
+  }
+  if (include_audio_level_indication_.load()) {
+    frame.audio_level_dbov = rms_level_.Average();
+  }
+  if (!rtp_sender_audio_->SendAudio(frame)) {
+    RTC_DLOG(LS_ERROR)
+        << "ChannelSend::SendData() failed to send data to RTP/RTCP module";
+    return -1;
+  }
+
+  return 0;
+}
+
+ChannelSend::ChannelSend(
+    Clock* clock,
+    TaskQueueFactory* task_queue_factory,
+    Transport* rtp_transport,
+    RtcpRttStats* rtcp_rtt_stats,
+    RtcEventLog* rtc_event_log,
+    FrameEncryptorInterface* frame_encryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    bool extmap_allow_mixed,
+    int rtcp_report_interval_ms,
+    uint32_t ssrc,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    RtpTransportControllerSendInterface* transport_controller,
+    const FieldTrialsView& field_trials)
+    : ssrc_(ssrc),
+      event_log_(rtc_event_log),
+      rtcp_counter_observer_(new RtcpCounterObserver(ssrc)),
+      rtp_packet_pacer_proxy_(new RtpPacketSenderProxy()),
+      retransmission_rate_limiter_(
+          new RateLimiter(clock, kMaxRetransmissionWindowMs)),
+      frame_encryptor_(frame_encryptor),
+      crypto_options_(crypto_options),
+      encoder_queue_(task_queue_factory->CreateTaskQueue(
+          "AudioEncoder",
+          TaskQueueFactory::Priority::NORMAL)) {
+  audio_coding_ = AudioCodingModule::Create();
+
+  RtpRtcpInterface::Configuration configuration;
+  configuration.report_block_data_observer = this;
+  configuration.network_link_rtcp_observer =
+      transport_controller->GetRtcpObserver();
+  configuration.transport_feedback_callback =
+      transport_controller->transport_feedback_observer();
+  configuration.clock = clock;
+  configuration.audio = true;
+  configuration.outgoing_transport = rtp_transport;
+
+  configuration.paced_sender = rtp_packet_pacer_proxy_.get();
+
+  configuration.event_log = event_log_;
+  configuration.rtt_stats = rtcp_rtt_stats;
+  configuration.rtcp_packet_type_counter_observer =
+      rtcp_counter_observer_.get();
+  if (field_trials.IsDisabled("WebRTC-DisableRtxRateLimiter")) {
+    configuration.retransmission_rate_limiter =
+        retransmission_rate_limiter_.get();
+  }
+  configuration.extmap_allow_mixed = extmap_allow_mixed;
+  configuration.rtcp_report_interval_ms = rtcp_report_interval_ms;
+  configuration.rtcp_packet_type_counter_observer = this;
+
+  configuration.local_media_ssrc = ssrc;
+
+  rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration);
+  rtp_rtcp_->SetSendingMediaStatus(false);
+
+  rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(configuration.clock,
+                                                       rtp_rtcp_->RtpSender());
+
+  // Ensure that RTCP is enabled by default for the created channel.
+  rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
+
+  int error = audio_coding_->RegisterTransportCallback(this);
+  RTC_DCHECK_EQ(0, error);
+}
+
+ChannelSend::~ChannelSend() {
+  RTC_DCHECK(construction_thread_.IsCurrent());
+
+  // Resets the delegate's callback to ChannelSend::SendRtpAudio.
+  if (frame_transformer_delegate_)
+    frame_transformer_delegate_->Reset();
+
+  StopSend();
+  int error = audio_coding_->RegisterTransportCallback(NULL);
+  RTC_DCHECK_EQ(0, error);
+}
+
+void ChannelSend::StartSend() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK(!sending_);
+  sending_ = true;
+
+  RTC_DCHECK(packet_router_);
+  packet_router_->AddSendRtpModule(rtp_rtcp_.get(), /*remb_candidate=*/false);
+  rtp_rtcp_->SetSendingMediaStatus(true);
+  int ret = rtp_rtcp_->SetSendingStatus(true);
+  RTC_DCHECK_EQ(0, ret);
+
+  // It is now OK to start processing on the encoder task queue.
+  first_frame_.store(true);
+  encoder_queue_is_active_.store(true);
+}
+
+void ChannelSend::StopSend() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!sending_) {
+    return;
+  }
+  sending_ = false;
+  encoder_queue_is_active_.store(false);
+
+  // Wait until all pending encode tasks are executed and clear any remaining
+  // buffers in the encoder.
+  rtc::Event flush;
+  encoder_queue_.PostTask([this, &flush]() {
+    RTC_DCHECK_RUN_ON(&encoder_queue_);
+    CallEncoder([](AudioEncoder* encoder) { encoder->Reset(); });
+    flush.Set();
+  });
+  flush.Wait(rtc::Event::kForever);
+
+  // Reset sending SSRC and sequence number and triggers direct transmission
+  // of RTCP BYE
+  if (rtp_rtcp_->SetSendingStatus(false) == -1) {
+    RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending";
+  }
+  rtp_rtcp_->SetSendingMediaStatus(false);
+
+  RTC_DCHECK(packet_router_);
+  packet_router_->RemoveSendRtpModule(rtp_rtcp_.get());
+  rtp_packet_pacer_proxy_->RemovePacketsForSsrc(rtp_rtcp_->SSRC());
+}
+
+void ChannelSend::SetEncoder(int payload_type,
+                             std::unique_ptr<AudioEncoder> encoder) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_GE(payload_type, 0);
+  RTC_DCHECK_LE(payload_type, 127);
+
+  // The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate)
+  // as well as some other things, so we collect this info and send it along.
+  rtp_rtcp_->RegisterSendPayloadFrequency(payload_type,
+                                          encoder->RtpTimestampRateHz());
+  rtp_sender_audio_->RegisterAudioPayload("audio", payload_type,
+                                          encoder->RtpTimestampRateHz(),
+                                          encoder->NumChannels(), 0);
+
+  audio_coding_->SetEncoder(std::move(encoder));
+}
+
+void ChannelSend::ModifyEncoder(
+    rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) {
+  // This method can be called on the worker thread, module process thread
+  // or network thread. Audio coding is thread safe, so we do not need to
+  // enforce the calling thread.
+  audio_coding_->ModifyEncoder(modifier);
+}
+
+void ChannelSend::CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) {
+  ModifyEncoder([modifier](std::unique_ptr<AudioEncoder>* encoder_ptr) {
+    if (*encoder_ptr) {
+      modifier(encoder_ptr->get());
+    } else {
+      RTC_DLOG(LS_WARNING) << "Trying to call unset encoder.";
+    }
+  });
+}
+
+void ChannelSend::OnBitrateAllocation(BitrateAllocationUpdate update) {
+  // This method can be called on the worker thread, module process thread
+  // or on a TaskQueue via VideoSendStreamImpl::OnEncoderConfigurationChanged.
+  // TODO(solenberg): Figure out a good way to check this or enforce calling
+  // rules.
+  // RTC_DCHECK(worker_thread_checker_.IsCurrent() ||
+  //            module_process_thread_checker_.IsCurrent());
+  CallEncoder([&](AudioEncoder* encoder) {
+    encoder->OnReceivedUplinkAllocation(update);
+  });
+  retransmission_rate_limiter_->SetMaxRate(update.target_bitrate.bps());
+}
+
+int ChannelSend::GetTargetBitrate() const {
+  return audio_coding_->GetTargetBitrate();
+}
+
+void ChannelSend::OnReportBlockDataUpdated(ReportBlockData report_block) {
+  float packet_loss_rate = report_block.fraction_lost();
+  CallEncoder([&](AudioEncoder* encoder) {
+    encoder->OnReceivedUplinkPacketLossFraction(packet_loss_rate);
+  });
+}
+
+void ChannelSend::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+
+  // Deliver RTCP packet to RTP/RTCP module for parsing
+  rtp_rtcp_->IncomingRtcpPacket(rtc::MakeArrayView(data, length));
+
+  int64_t rtt = GetRTT();
+  if (rtt == 0) {
+    // Waiting for valid RTT.
+    return;
+  }
+
+  int64_t nack_window_ms = rtt;
+  if (nack_window_ms < kMinRetransmissionWindowMs) {
+    nack_window_ms = kMinRetransmissionWindowMs;
+  } else if (nack_window_ms > kMaxRetransmissionWindowMs) {
+    nack_window_ms = kMaxRetransmissionWindowMs;
+  }
+  retransmission_rate_limiter_->SetWindowSize(nack_window_ms);
+
+  OnReceivedRtt(rtt);
+}
+
+void ChannelSend::SetInputMute(bool enable) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  MutexLock lock(&volume_settings_mutex_);
+  input_mute_ = enable;
+}
+
+bool ChannelSend::InputMute() const {
+  MutexLock lock(&volume_settings_mutex_);
+  return input_mute_;
+}
+
+bool ChannelSend::SendTelephoneEventOutband(int event, int duration_ms) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_LE(0, event);
+  RTC_DCHECK_GE(255, event);
+  RTC_DCHECK_LE(0, duration_ms);
+  RTC_DCHECK_GE(65535, duration_ms);
+  if (!sending_) {
+    return false;
+  }
+  if (rtp_sender_audio_->SendTelephoneEvent(
+          event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
+    RTC_DLOG(LS_ERROR) << "SendTelephoneEvent() failed to send event";
+    return false;
+  }
+  return true;
+}
+
+void ChannelSend::RegisterCngPayloadType(int payload_type,
+                                         int payload_frequency) {
+  rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, payload_frequency);
+  rtp_sender_audio_->RegisterAudioPayload("CN", payload_type, payload_frequency,
+                                          1, 0);
+}
+
+void ChannelSend::SetSendTelephoneEventPayloadType(int payload_type,
+                                                   int payload_frequency) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_LE(0, payload_type);
+  RTC_DCHECK_GE(127, payload_type);
+  rtp_rtcp_->RegisterSendPayloadFrequency(payload_type, payload_frequency);
+  rtp_sender_audio_->RegisterAudioPayload("telephone-event", payload_type,
+                                          payload_frequency, 0, 0);
+}
+
+void ChannelSend::SetSendAudioLevelIndicationStatus(bool enable, int id) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  include_audio_level_indication_.store(enable);
+  if (enable) {
+    rtp_rtcp_->RegisterRtpHeaderExtension(webrtc::AudioLevel::Uri(), id);
+  } else {
+    rtp_rtcp_->DeregisterSendRtpHeaderExtension(webrtc::AudioLevel::Uri());
+  }
+}
+
+void ChannelSend::RegisterSenderCongestionControlObjects(
+    RtpTransportControllerSendInterface* transport) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RtpPacketSender* rtp_packet_pacer = transport->packet_sender();
+  PacketRouter* packet_router = transport->packet_router();
+
+  RTC_DCHECK(rtp_packet_pacer);
+  RTC_DCHECK(packet_router);
+  RTC_DCHECK(!packet_router_);
+  rtp_packet_pacer_proxy_->SetPacketPacer(rtp_packet_pacer);
+  rtp_rtcp_->SetStorePacketsStatus(true, 600);
+  packet_router_ = packet_router;
+}
+
+void ChannelSend::ResetSenderCongestionControlObjects() {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK(packet_router_);
+  rtp_rtcp_->SetStorePacketsStatus(false, 600);
+  packet_router_ = nullptr;
+  rtp_packet_pacer_proxy_->SetPacketPacer(nullptr);
+}
+
+void ChannelSend::SetRTCP_CNAME(absl::string_view c_name) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // Note: SetCNAME() accepts a c string of length at most 255.
+  const std::string c_name_limited(c_name.substr(0, 255));
+  int ret = rtp_rtcp_->SetCNAME(c_name_limited.c_str()) != 0;
+  RTC_DCHECK_EQ(0, ret) << "SetRTCP_CNAME() failed to set RTCP CNAME";
+}
+
+std::vector<ReportBlockData> ChannelSend::GetRemoteRTCPReportBlocks() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  // Get the report blocks from the latest received RTCP Sender or Receiver
+  // Report. Each element in the vector contains the sender's SSRC and a
+  // report block according to RFC 3550.
+  return rtp_rtcp_->GetLatestReportBlockData();
+}
+
+CallSendStatistics ChannelSend::GetRTCPStatistics() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  CallSendStatistics stats = {0};
+  stats.rttMs = GetRTT();
+  stats.rtcp_packet_type_counts = rtcp_counter_observer_->GetCounts();
+
+  StreamDataCounters rtp_stats;
+  StreamDataCounters rtx_stats;
+  rtp_rtcp_->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
+  stats.payload_bytes_sent =
+      rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
+  stats.header_and_padding_bytes_sent =
+      rtp_stats.transmitted.padding_bytes + rtp_stats.transmitted.header_bytes +
+      rtx_stats.transmitted.padding_bytes + rtx_stats.transmitted.header_bytes;
+
+  // TODO(https://crbug.com/webrtc/10555): RTX retransmissions should show up in
+  // separate outbound-rtp stream objects.
+  stats.retransmitted_bytes_sent = rtp_stats.retransmitted.payload_bytes;
+  stats.packetsSent =
+      rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
+  stats.total_packet_send_delay = rtp_stats.transmitted.total_packet_delay;
+  stats.retransmitted_packets_sent = rtp_stats.retransmitted.packets;
+  stats.report_block_datas = rtp_rtcp_->GetLatestReportBlockData();
+
+  {
+    MutexLock lock(&rtcp_counter_mutex_);
+    stats.nacks_received = rtcp_packet_type_counter_.nack_packets;
+  }
+
+  return stats;
+}
+
+void ChannelSend::RtcpPacketTypesCounterUpdated(
+    uint32_t ssrc,
+    const RtcpPacketTypeCounter& packet_counter) {
+  if (ssrc != ssrc_) {
+    return;
+  }
+  MutexLock lock(&rtcp_counter_mutex_);
+  rtcp_packet_type_counter_ = packet_counter;
+}
+
+void ChannelSend::ProcessAndEncodeAudio(
+    std::unique_ptr<AudioFrame> audio_frame) {
+  TRACE_EVENT0("webrtc", "ChannelSend::ProcessAndEncodeAudio");
+
+  RTC_DCHECK_RUNS_SERIALIZED(&audio_thread_race_checker_);
+  RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
+  RTC_DCHECK_LE(audio_frame->num_channels_, 8);
+
+  if (!encoder_queue_is_active_.load()) {
+    return;
+  }
+
+  // Update `timestamp_` based on the capture timestamp for the first frame
+  // after sending is resumed.
+  if (first_frame_.load()) {
+    first_frame_.store(false);
+    if (last_capture_timestamp_ms_ &&
+        audio_frame->absolute_capture_timestamp_ms()) {
+      int64_t diff_ms = *audio_frame->absolute_capture_timestamp_ms() -
+                        *last_capture_timestamp_ms_;
+      // Truncate to whole frames and subtract one since `timestamp_` was
+      // incremented after the last frame.
+      int64_t diff_frames = diff_ms * audio_frame->sample_rate_hz() / 1000 /
+                                audio_frame->samples_per_channel() -
+                            1;
+      timestamp_ += std::max<int64_t>(
+          diff_frames * audio_frame->samples_per_channel(), 0);
+    }
+  }
+
+  audio_frame->timestamp_ = timestamp_;
+  timestamp_ += audio_frame->samples_per_channel_;
+  last_capture_timestamp_ms_ = audio_frame->absolute_capture_timestamp_ms();
+
+  // Profile time between when the audio frame is added to the task queue and
+  // when the task is actually executed.
+  audio_frame->UpdateProfileTimeStamp();
+  encoder_queue_.PostTask(
+      [this, audio_frame = std::move(audio_frame)]() mutable {
+        RTC_DCHECK_RUN_ON(&encoder_queue_);
+        if (!encoder_queue_is_active_.load()) {
+          return;
+        }
+        // Measure time between when the audio frame is added to the task queue
+        // and when the task is actually executed. Goal is to keep track of
+        // unwanted extra latency added by the task queue.
+        RTC_HISTOGRAM_COUNTS_10000("WebRTC.Audio.EncodingTaskQueueLatencyMs",
+                                   audio_frame->ElapsedProfileTimeMs());
+
+        bool is_muted = InputMute();
+        AudioFrameOperations::Mute(audio_frame.get(), previous_frame_muted_,
+                                   is_muted);
+
+        if (include_audio_level_indication_.load()) {
+          size_t length =
+              audio_frame->samples_per_channel_ * audio_frame->num_channels_;
+          RTC_CHECK_LE(length, AudioFrame::kMaxDataSizeBytes);
+          if (is_muted && previous_frame_muted_) {
+            rms_level_.AnalyzeMuted(length);
+          } else {
+            rms_level_.Analyze(
+                rtc::ArrayView<const int16_t>(audio_frame->data(), length));
+          }
+        }
+        previous_frame_muted_ = is_muted;
+
+        // This call will trigger AudioPacketizationCallback::SendData if
+        // encoding is done and payload is ready for packetization and
+        // transmission. Otherwise, it will return without invoking the
+        // callback.
+        if (audio_coding_->Add10MsData(*audio_frame) < 0) {
+          RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
+          return;
+        }
+      });
+}
+
+ANAStats ChannelSend::GetANAStatistics() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return audio_coding_->GetANAStats();
+}
+
+RtpRtcpInterface* ChannelSend::GetRtpRtcp() const {
+  return rtp_rtcp_.get();
+}
+
+int64_t ChannelSend::GetRTT() const {
+  std::vector<ReportBlockData> report_blocks =
+      rtp_rtcp_->GetLatestReportBlockData();
+  if (report_blocks.empty()) {
+    return 0;
+  }
+
+  // We don't know in advance the remote ssrc used by the other end's receiver
+  // reports, so use the first report block for the RTT.
+  return report_blocks.front().last_rtt().ms();
+}
+
+void ChannelSend::SetFrameEncryptor(
+    rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  encoder_queue_.PostTask([this, frame_encryptor]() mutable {
+    RTC_DCHECK_RUN_ON(&encoder_queue_);
+    frame_encryptor_ = std::move(frame_encryptor);
+  });
+}
+
+void ChannelSend::SetEncoderToPacketizerFrameTransformer(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  if (!frame_transformer)
+    return;
+
+  encoder_queue_.PostTask(
+      [this, frame_transformer = std::move(frame_transformer)]() mutable {
+        RTC_DCHECK_RUN_ON(&encoder_queue_);
+        InitFrameTransformerDelegate(std::move(frame_transformer));
+      });
+}
+
+void ChannelSend::OnReceivedRtt(int64_t rtt_ms) {
+  // Invoke audio encoders OnReceivedRtt().
+  CallEncoder(
+      [rtt_ms](AudioEncoder* encoder) { encoder->OnReceivedRtt(rtt_ms); });
+}
+
+void ChannelSend::InitFrameTransformerDelegate(
+    rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(&encoder_queue_);
+  RTC_DCHECK(frame_transformer);
+  RTC_DCHECK(!frame_transformer_delegate_);
+
+  // Pass a callback to ChannelSend::SendRtpAudio, to be called by the delegate
+  // to send the transformed audio.
+  ChannelSendFrameTransformerDelegate::SendFrameCallback send_audio_callback =
+      [this](AudioFrameType frameType, uint8_t payloadType,
+             uint32_t rtp_timestamp_with_offset,
+             rtc::ArrayView<const uint8_t> payload,
+             int64_t absolute_capture_timestamp_ms) {
+        RTC_DCHECK_RUN_ON(&encoder_queue_);
+        return SendRtpAudio(
+            frameType, payloadType,
+            rtp_timestamp_with_offset - rtp_rtcp_->StartTimestamp(), payload,
+            absolute_capture_timestamp_ms);
+      };
+  frame_transformer_delegate_ =
+      rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
+          std::move(send_audio_callback), std::move(frame_transformer),
+          &encoder_queue_);
+  frame_transformer_delegate_->Init();
+}
+
+}  // namespace
+
+std::unique_ptr<ChannelSendInterface> CreateChannelSend(
+    Clock* clock,
+    TaskQueueFactory* task_queue_factory,
+    Transport* rtp_transport,
+    RtcpRttStats* rtcp_rtt_stats,
+    RtcEventLog* rtc_event_log,
+    FrameEncryptorInterface* frame_encryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    bool extmap_allow_mixed,
+    int rtcp_report_interval_ms,
+    uint32_t ssrc,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    RtpTransportControllerSendInterface* transport_controller,
+    const FieldTrialsView& field_trials) {
+  return std::make_unique<ChannelSend>(
+      clock, task_queue_factory, rtp_transport, rtcp_rtt_stats, rtc_event_log,
+      frame_encryptor, crypto_options, extmap_allow_mixed,
+      rtcp_report_interval_ms, ssrc, std::move(frame_transformer),
+      transport_controller, field_trials);
+}
+
+}  // namespace voe
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/channel_send.h b/third_party/libwebrtc/audio/channel_send.h
new file mode 100644
index 0000000000..f0c9232296
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_send.h
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_CHANNEL_SEND_H_
+#define AUDIO_CHANNEL_SEND_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/crypto/crypto_options.h"
+#include "api/field_trials_view.h"
+#include "api/frame_transformer_interface.h"
+#include "api/function_view.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
+
+namespace webrtc {
+
+class FrameEncryptorInterface;
+class RtcEventLog;
+class RtpTransportControllerSendInterface;
+
+struct CallSendStatistics {
+  int64_t rttMs;
+  int64_t payload_bytes_sent;
+  int64_t header_and_padding_bytes_sent;
+  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedbytessent
+  uint64_t retransmitted_bytes_sent;
+  int packetsSent;
+  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
+  TimeDelta total_packet_send_delay = TimeDelta::Zero();
+  // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-retransmittedpacketssent
+  uint64_t retransmitted_packets_sent;
+  RtcpPacketTypeCounter rtcp_packet_type_counts;
+  // A snapshot of Report Blocks with additional data of interest to statistics.
+  // Within this list, the sender-source SSRC pair is unique and per-pair the
+  // ReportBlockData represents the latest Report Block that was received for
+  // that pair.
+  std::vector<ReportBlockData> report_block_datas;
+  uint32_t nacks_received;
+};
+
+namespace voe {
+
+class ChannelSendInterface {
+ public:
+  virtual ~ChannelSendInterface() = default;
+
+  virtual void ReceivedRTCPPacket(const uint8_t* packet, size_t length) = 0;
+
+  virtual CallSendStatistics GetRTCPStatistics() const = 0;
+
+  virtual void SetEncoder(int payload_type,
+                          std::unique_ptr<AudioEncoder> encoder) = 0;
+  virtual void ModifyEncoder(
+      rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier) = 0;
+  virtual void CallEncoder(rtc::FunctionView<void(AudioEncoder*)> modifier) = 0;
+
+  // Use 0 to indicate that the extension should not be registered.
+  virtual void SetRTCP_CNAME(absl::string_view c_name) = 0;
+  virtual void SetSendAudioLevelIndicationStatus(bool enable, int id) = 0;
+  virtual void RegisterSenderCongestionControlObjects(
+      RtpTransportControllerSendInterface* transport) = 0;
+  virtual void ResetSenderCongestionControlObjects() = 0;
+  virtual std::vector<ReportBlockData> GetRemoteRTCPReportBlocks() const = 0;
+  virtual ANAStats GetANAStatistics() const = 0;
+  virtual void RegisterCngPayloadType(int payload_type,
+                                      int payload_frequency) = 0;
+  virtual void SetSendTelephoneEventPayloadType(int payload_type,
+                                                int payload_frequency) = 0;
+  virtual bool SendTelephoneEventOutband(int event, int duration_ms) = 0;
+  virtual void OnBitrateAllocation(BitrateAllocationUpdate update) = 0;
+  virtual int GetTargetBitrate() const = 0;
+  virtual void SetInputMute(bool muted) = 0;
+
+  virtual void ProcessAndEncodeAudio(
+      std::unique_ptr<AudioFrame> audio_frame) = 0;
+  virtual RtpRtcpInterface* GetRtpRtcp() const = 0;
+
+  // In RTP we currently rely on RTCP packets (`ReceivedRTCPPacket`) to inform
+  // about RTT.
+  // In media transport we rely on the TargetTransferRateObserver instead.
+  // In other words, if you are using RTP, you should expect
+  // `ReceivedRTCPPacket` to be called, if you are using media transport,
+  // `OnTargetTransferRate` will be called.
+  //
+  // In future, RTP media will move to the media transport implementation and
+  // these conditions will be removed.
+  // Returns the RTT in milliseconds.
+  virtual int64_t GetRTT() const = 0;
+  virtual void StartSend() = 0;
+  virtual void StopSend() = 0;
+
+  // E2EE Custom Audio Frame Encryption (Optional)
+  virtual void SetFrameEncryptor(
+      rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0;
+
+  // Sets a frame transformer between encoder and packetizer, to transform
+  // encoded frames before sending them out the network.
+  virtual void SetEncoderToPacketizerFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+          frame_transformer) = 0;
+};
+
+std::unique_ptr<ChannelSendInterface> CreateChannelSend(
+    Clock* clock,
+    TaskQueueFactory* task_queue_factory,
+    Transport* rtp_transport,
+    RtcpRttStats* rtcp_rtt_stats,
+    RtcEventLog* rtc_event_log,
+    FrameEncryptorInterface* frame_encryptor,
+    const webrtc::CryptoOptions& crypto_options,
+    bool extmap_allow_mixed,
+    int rtcp_report_interval_ms,
+    uint32_t ssrc,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    RtpTransportControllerSendInterface* transport_controller,
+    const FieldTrialsView& field_trials);
+
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // AUDIO_CHANNEL_SEND_H_
diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc
new file mode 100644
index 0000000000..0f85216e92
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.cc
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/channel_send_frame_transformer_delegate.h"
+
+#include <utility>
+
+namespace webrtc {
+namespace {
+
+using IfaceFrameType = TransformableAudioFrameInterface::FrameType;
+
+IfaceFrameType InternalFrameTypeToInterfaceFrameType(
+    const AudioFrameType frame_type) {
+  switch (frame_type) {
+    case AudioFrameType::kEmptyFrame:
+      return IfaceFrameType::kEmptyFrame;
+    case AudioFrameType::kAudioFrameSpeech:
+      return IfaceFrameType::kAudioFrameSpeech;
+    case AudioFrameType::kAudioFrameCN:
+      return IfaceFrameType::kAudioFrameCN;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return IfaceFrameType::kEmptyFrame;
+}
+
+AudioFrameType InterfaceFrameTypeToInternalFrameType(
+    const IfaceFrameType frame_type) {
+  switch (frame_type) {
+    case IfaceFrameType::kEmptyFrame:
+      return AudioFrameType::kEmptyFrame;
+    case IfaceFrameType::kAudioFrameSpeech:
+      return AudioFrameType::kAudioFrameSpeech;
+    case IfaceFrameType::kAudioFrameCN:
+      return AudioFrameType::kAudioFrameCN;
+  }
+  RTC_DCHECK_NOTREACHED();
+  return AudioFrameType::kEmptyFrame;
+}
+
+class TransformableOutgoingAudioFrame
+    : public TransformableAudioFrameInterface {
+ public:
+  TransformableOutgoingAudioFrame(
+      AudioFrameType frame_type,
+      uint8_t payload_type,
+      uint32_t rtp_timestamp_with_offset,
+      const uint8_t* payload_data,
+      size_t payload_size,
+      absl::optional<uint64_t> absolute_capture_timestamp_ms,
+      uint32_t ssrc)
+      : frame_type_(frame_type),
+        payload_type_(payload_type),
+        rtp_timestamp_with_offset_(rtp_timestamp_with_offset),
+        payload_(payload_data, payload_size),
+        absolute_capture_timestamp_ms_(absolute_capture_timestamp_ms),
+        ssrc_(ssrc) {}
+  ~TransformableOutgoingAudioFrame() override = default;
+  rtc::ArrayView<const uint8_t> GetData() const override { return payload_; }
+  void SetData(rtc::ArrayView<const uint8_t> data) override {
+    payload_.SetData(data.data(), data.size());
+  }
+  uint32_t GetTimestamp() const override { return rtp_timestamp_with_offset_; }
+  uint32_t GetSsrc() const override { return ssrc_; }
+
+  IfaceFrameType Type() const override {
+    return InternalFrameTypeToInterfaceFrameType(frame_type_);
+  }
+
+  uint8_t GetPayloadType() const override { return payload_type_; }
+  Direction GetDirection() const override { return Direction::kSender; }
+
+  rtc::ArrayView<const uint32_t> GetContributingSources() const override {
+    return {};
+  }
+
+  const absl::optional<uint16_t> SequenceNumber() const override {
+    return absl::nullopt;
+  }
+
+  void SetRTPTimestamp(uint32_t rtp_timestamp_with_offset) override {
+    rtp_timestamp_with_offset_ = rtp_timestamp_with_offset;
+  }
+
+  absl::optional<uint64_t> AbsoluteCaptureTimestamp() const override {
+    return absolute_capture_timestamp_ms_;
+  }
+
+ private:
+  AudioFrameType frame_type_;
+  uint8_t payload_type_;
+  uint32_t rtp_timestamp_with_offset_;
+  rtc::Buffer payload_;
+  absl::optional<uint64_t> absolute_capture_timestamp_ms_;
+  uint32_t ssrc_;
+};
+}  // namespace
+
+ChannelSendFrameTransformerDelegate::ChannelSendFrameTransformerDelegate(
+    SendFrameCallback send_frame_callback,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+    rtc::TaskQueue* encoder_queue)
+    : send_frame_callback_(send_frame_callback),
+      frame_transformer_(std::move(frame_transformer)),
+      encoder_queue_(encoder_queue) {}
+
+void ChannelSendFrameTransformerDelegate::Init() {
+  frame_transformer_->RegisterTransformedFrameCallback(
+      rtc::scoped_refptr<TransformedFrameCallback>(this));
+}
+
+void ChannelSendFrameTransformerDelegate::Reset() {
+  frame_transformer_->UnregisterTransformedFrameCallback();
+  frame_transformer_ = nullptr;
+
+  MutexLock lock(&send_lock_);
+  send_frame_callback_ = SendFrameCallback();
+}
+
+void ChannelSendFrameTransformerDelegate::Transform(
+    AudioFrameType frame_type,
+    uint8_t payload_type,
+    uint32_t rtp_timestamp,
+    const uint8_t* payload_data,
+    size_t payload_size,
+    int64_t absolute_capture_timestamp_ms,
+    uint32_t ssrc) {
+  frame_transformer_->Transform(
+      std::make_unique<TransformableOutgoingAudioFrame>(
+          frame_type, payload_type, rtp_timestamp, payload_data, payload_size,
+          absolute_capture_timestamp_ms, ssrc));
+}
+
+void ChannelSendFrameTransformerDelegate::OnTransformedFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) {
+  MutexLock lock(&send_lock_);
+  if (!send_frame_callback_)
+    return;
+  rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate(this);
+  encoder_queue_->PostTask(
+      [delegate = std::move(delegate), frame = std::move(frame)]() mutable {
+        delegate->SendFrame(std::move(frame));
+      });
+}
+
+void ChannelSendFrameTransformerDelegate::SendFrame(
+    std::unique_ptr<TransformableFrameInterface> frame) const {
+  MutexLock lock(&send_lock_);
+  RTC_DCHECK_RUN_ON(encoder_queue_);
+  if (!send_frame_callback_)
+    return;
+  auto* transformed_frame =
+      static_cast<TransformableAudioFrameInterface*>(frame.get());
+  send_frame_callback_(
+      InterfaceFrameTypeToInternalFrameType(transformed_frame->Type()),
+      transformed_frame->GetPayloadType(), transformed_frame->GetTimestamp(),
+      transformed_frame->GetData(),
+      transformed_frame->AbsoluteCaptureTimestamp()
+          ? *transformed_frame->AbsoluteCaptureTimestamp()
+          : 0);
+}
+
+std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
+    TransformableAudioFrameInterface* original) {
+  return std::make_unique<TransformableOutgoingAudioFrame>(
+      InterfaceFrameTypeToInternalFrameType(original->Type()),
+      original->GetPayloadType(), original->GetTimestamp(),
+      original->GetData().data(), original->GetData().size(),
+      original->AbsoluteCaptureTimestamp(), original->GetSsrc());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h
new file mode 100644
index 0000000000..eb0027e4c8
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate.h
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_
+#define AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_
+
+#include <memory>
+
+#include "api/frame_transformer_interface.h"
+#include "api/sequence_checker.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+
+namespace webrtc {
+
+// Delegates calls to FrameTransformerInterface to transform frames, and to
+// ChannelSend to send the transformed frames using `send_frame_callback_` on
+// the `encoder_queue_`.
+// OnTransformedFrame() can be called from any thread, the delegate ensures
+// thread-safe access to the ChannelSend callback.
+class ChannelSendFrameTransformerDelegate : public TransformedFrameCallback {
+ public:
+  using SendFrameCallback =
+      std::function<int32_t(AudioFrameType frameType,
+                            uint8_t payloadType,
+                            uint32_t rtp_timestamp_with_offset,
+                            rtc::ArrayView<const uint8_t> payload,
+                            int64_t absolute_capture_timestamp_ms)>;
+  ChannelSendFrameTransformerDelegate(
+      SendFrameCallback send_frame_callback,
+      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
+      rtc::TaskQueue* encoder_queue);
+
+  // Registers `this` as callback for `frame_transformer_`, to get the
+  // transformed frames.
+  void Init();
+
+  // Unregisters and releases the `frame_transformer_` reference, and resets
+  // `send_frame_callback_` under lock. Called from ChannelSend destructor to
+  // prevent running the callback on a dangling channel.
+  void Reset();
+
+  // Delegates the call to FrameTransformerInterface::TransformFrame, to
+  // transform the frame asynchronously.
+  void Transform(AudioFrameType frame_type,
+                 uint8_t payload_type,
+                 uint32_t rtp_timestamp,
+                 const uint8_t* payload_data,
+                 size_t payload_size,
+                 int64_t absolute_capture_timestamp_ms,
+                 uint32_t ssrc);
+
+  // Implements TransformedFrameCallback. Can be called on any thread.
+  void OnTransformedFrame(
+      std::unique_ptr<TransformableFrameInterface> frame) override;
+
+  // Delegates the call to ChannelSend::SendRtpAudio on the `encoder_queue_`,
+  // by calling `send_audio_callback_`.
+  void SendFrame(std::unique_ptr<TransformableFrameInterface> frame) const;
+
+ protected:
+  ~ChannelSendFrameTransformerDelegate() override = default;
+
+ private:
+  mutable Mutex send_lock_;
+  SendFrameCallback send_frame_callback_ RTC_GUARDED_BY(send_lock_);
+  rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_;
+  rtc::TaskQueue* encoder_queue_ RTC_GUARDED_BY(send_lock_);
+};
+
+std::unique_ptr<TransformableAudioFrameInterface> CloneSenderAudioFrame(
+    TransformableAudioFrameInterface* original);
+
+}  // namespace webrtc
+#endif  // AUDIO_CHANNEL_SEND_FRAME_TRANSFORMER_DELEGATE_H_
diff --git a/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate_unittest.cc b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate_unittest.cc
new file mode 100644
index 0000000000..f75d4a8ab7
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_send_frame_transformer_delegate_unittest.cc
@@ -0,0 +1,172 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/channel_send_frame_transformer_delegate.h"
+
+#include <memory>
+#include <utility>
+
+#include "rtc_base/task_queue_for_test.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_frame_transformer.h"
+#include "test/mock_transformable_frame.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::SaveArg;
+
+class MockChannelSend {
+ public:
+  MockChannelSend() = default;
+  ~MockChannelSend() = default;
+
+  MOCK_METHOD(int32_t,
+              SendFrame,
+              (AudioFrameType frameType,
+               uint8_t payloadType,
+               uint32_t rtp_timestamp,
+               rtc::ArrayView<const uint8_t> payload,
+               int64_t absolute_capture_timestamp_ms));
+
+  ChannelSendFrameTransformerDelegate::SendFrameCallback callback() {
+    return [this](AudioFrameType frameType, uint8_t payloadType,
+                  uint32_t rtp_timestamp, rtc::ArrayView<const uint8_t> payload,
+                  int64_t absolute_capture_timestamp_ms) {
+      return SendFrame(frameType, payloadType, rtp_timestamp, payload,
+                       absolute_capture_timestamp_ms);
+    };
+  }
+};
+
+std::unique_ptr<MockTransformableAudioFrame> CreateMockReceiverFrame() {
+  const uint8_t mock_data[] = {1, 2, 3, 4};
+  std::unique_ptr<MockTransformableAudioFrame> mock_frame =
+      std::make_unique<MockTransformableAudioFrame>();
+  rtc::ArrayView<const uint8_t> payload(mock_data);
+  ON_CALL(*mock_frame, GetData).WillByDefault(Return(payload));
+  ON_CALL(*mock_frame, GetPayloadType).WillByDefault(Return(0));
+  ON_CALL(*mock_frame, GetDirection)
+      .WillByDefault(Return(TransformableFrameInterface::Direction::kReceiver));
+  return mock_frame;
+}
+
+// Test that the delegate registers itself with the frame transformer on Init().
+TEST(ChannelSendFrameTransformerDelegateTest,
+     RegisterTransformedFrameCallbackOnInit) {
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<MockFrameTransformer>();
+  rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
+          ChannelSendFrameTransformerDelegate::SendFrameCallback(),
+          mock_frame_transformer, nullptr);
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback);
+  delegate->Init();
+}
+
+// Test that the delegate unregisters itself from the frame transformer on
+// Reset().
+TEST(ChannelSendFrameTransformerDelegateTest,
+     UnregisterTransformedFrameCallbackOnReset) {
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<MockFrameTransformer>();
+  rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
+          ChannelSendFrameTransformerDelegate::SendFrameCallback(),
+          mock_frame_transformer, nullptr);
+  EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
+  delegate->Reset();
+}
+
+// Test that when the delegate receives a transformed frame from the frame
+// transformer, it passes it to the channel using the SendFrameCallback.
+TEST(ChannelSendFrameTransformerDelegateTest,
+     TransformRunsChannelSendCallback) {
+  TaskQueueForTest channel_queue("channel_queue");
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  MockChannelSend mock_channel;
+  rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
+          mock_channel.callback(), mock_frame_transformer, &channel_queue);
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  delegate->Init();
+  ASSERT_TRUE(callback);
+
+  const uint8_t data[] = {1, 2, 3, 4};
+  EXPECT_CALL(mock_channel, SendFrame);
+  ON_CALL(*mock_frame_transformer, Transform)
+      .WillByDefault(
+          [&callback](std::unique_ptr<TransformableFrameInterface> frame) {
+            callback->OnTransformedFrame(std::move(frame));
+          });
+  delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, data, sizeof(data), 0,
+                      0);
+  channel_queue.WaitForPreviouslyPostedTasks();
+}
+
+// Test that when the delegate receives a Incoming frame from the frame
+// transformer, it passes it to the channel using the SendFrameCallback.
+TEST(ChannelSendFrameTransformerDelegateTest,
+     TransformRunsChannelSendCallbackForIncomingFrame) {
+  TaskQueueForTest channel_queue("channel_queue");
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<NiceMock<MockFrameTransformer>>();
+  MockChannelSend mock_channel;
+  rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
+          mock_channel.callback(), mock_frame_transformer, &channel_queue);
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  delegate->Init();
+  ASSERT_TRUE(callback);
+
+  const uint8_t data[] = {1, 2, 3, 4};
+  EXPECT_CALL(mock_channel, SendFrame).Times(0);
+  EXPECT_CALL(mock_channel, SendFrame(_, 0, 0, ElementsAre(1, 2, 3, 4), _));
+  ON_CALL(*mock_frame_transformer, Transform)
+      .WillByDefault(
+          [&callback](std::unique_ptr<TransformableFrameInterface> frame) {
+            callback->OnTransformedFrame(CreateMockReceiverFrame());
+          });
+  delegate->Transform(AudioFrameType::kEmptyFrame, 0, 0, data, sizeof(data), 0,
+                      0);
+  channel_queue.WaitForPreviouslyPostedTasks();
+}
+
+// Test that if the delegate receives a transformed frame after it has been
+// reset, it does not run the SendFrameCallback, as the channel is destroyed
+// after resetting the delegate.
+TEST(ChannelSendFrameTransformerDelegateTest,
+     OnTransformedDoesNotRunChannelSendCallbackAfterReset) {
+  TaskQueueForTest channel_queue("channel_queue");
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<testing::NiceMock<MockFrameTransformer>>();
+  MockChannelSend mock_channel;
+  rtc::scoped_refptr<ChannelSendFrameTransformerDelegate> delegate =
+      rtc::make_ref_counted<ChannelSendFrameTransformerDelegate>(
+          mock_channel.callback(), mock_frame_transformer, &channel_queue);
+
+  delegate->Reset();
+  EXPECT_CALL(mock_channel, SendFrame).Times(0);
+  delegate->OnTransformedFrame(std::make_unique<MockTransformableAudioFrame>());
+  channel_queue.WaitForPreviouslyPostedTasks();
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/channel_send_unittest.cc b/third_party/libwebrtc/audio/channel_send_unittest.cc
new file mode 100644
index 0000000000..b9406e1523
--- /dev/null
+++ b/third_party/libwebrtc/audio/channel_send_unittest.cc
@@ -0,0 +1,193 @@
+/*
+ *  Copyright 2023 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/channel_send.h"
+
+#include <utility>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/scoped_refptr.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "call/rtp_transport_controller_send.h"
+#include "rtc_base/gunit.h"
+#include "test/gtest.h"
+#include "test/mock_frame_transformer.h"
+#include "test/mock_transport.h"
+#include "test/scoped_key_value_config.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace voe {
+namespace {
+
+using ::testing::Invoke;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::SaveArg;
+
+constexpr int kRtcpIntervalMs = 1000;
+constexpr int kSsrc = 333;
+constexpr int kPayloadType = 1;
+constexpr int kSampleRateHz = 48000;
+constexpr int kRtpRateHz = 48000;
+
+BitrateConstraints GetBitrateConfig() {
+  BitrateConstraints bitrate_config;
+  bitrate_config.min_bitrate_bps = 10000;
+  bitrate_config.start_bitrate_bps = 100000;
+  bitrate_config.max_bitrate_bps = 1000000;
+  return bitrate_config;
+}
+
+class ChannelSendTest : public ::testing::Test {
+ protected:
+  ChannelSendTest()
+      : time_controller_(Timestamp::Seconds(1)),
+        transport_controller_(
+            time_controller_.GetClock(),
+            RtpTransportConfig{
+                .bitrate_config = GetBitrateConfig(),
+                .event_log = &event_log_,
+                .task_queue_factory = time_controller_.GetTaskQueueFactory(),
+                .trials = &field_trials_,
+            }) {
+    channel_ = voe::CreateChannelSend(
+        time_controller_.GetClock(), time_controller_.GetTaskQueueFactory(),
+        &transport_, nullptr, &event_log_, nullptr, crypto_options_, false,
+        kRtcpIntervalMs, kSsrc, nullptr, &transport_controller_, field_trials_);
+    encoder_factory_ = CreateBuiltinAudioEncoderFactory();
+    std::unique_ptr<AudioEncoder> encoder = encoder_factory_->MakeAudioEncoder(
+        kPayloadType, SdpAudioFormat("opus", kRtpRateHz, 2), {});
+    channel_->SetEncoder(kPayloadType, std::move(encoder));
+    transport_controller_.EnsureStarted();
+    channel_->RegisterSenderCongestionControlObjects(&transport_controller_);
+    ON_CALL(transport_, SendRtcp).WillByDefault(Return(true));
+    ON_CALL(transport_, SendRtp).WillByDefault(Return(true));
+  }
+
+  std::unique_ptr<AudioFrame> CreateAudioFrame() {
+    auto frame = std::make_unique<AudioFrame>();
+    frame->sample_rate_hz_ = kSampleRateHz;
+    frame->samples_per_channel_ = kSampleRateHz / 100;
+    frame->num_channels_ = 1;
+    frame->set_absolute_capture_timestamp_ms(
+        time_controller_.GetClock()->TimeInMilliseconds());
+    return frame;
+  }
+
+  void ProcessNextFrame() {
+    channel_->ProcessAndEncodeAudio(CreateAudioFrame());
+    // Advance time to process the task queue.
+    time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  }
+
+  GlobalSimulatedTimeController time_controller_;
+  webrtc::test::ScopedKeyValueConfig field_trials_;
+  RtcEventLogNull event_log_;
+  NiceMock<MockTransport> transport_;
+  CryptoOptions crypto_options_;
+  RtpTransportControllerSend transport_controller_;
+  std::unique_ptr<ChannelSendInterface> channel_;
+  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
+};
+
+TEST_F(ChannelSendTest, StopSendShouldResetEncoder) {
+  channel_->StartSend();
+  // Insert two frames which should trigger a new packet.
+  EXPECT_CALL(transport_, SendRtp).Times(1);
+  ProcessNextFrame();
+  ProcessNextFrame();
+
+  EXPECT_CALL(transport_, SendRtp).Times(0);
+  ProcessNextFrame();
+  // StopSend should clear the previous audio frame stored in the encoder.
+  channel_->StopSend();
+  channel_->StartSend();
+  // The following frame should not trigger a new packet since the encoder
+  // needs 20 ms audio.
+  EXPECT_CALL(transport_, SendRtp).Times(0);
+  ProcessNextFrame();
+}
+
+TEST_F(ChannelSendTest, IncreaseRtpTimestampByPauseDuration) {
+  channel_->StartSend();
+  uint32_t timestamp;
+  int sent_packets = 0;
+  auto send_rtp = [&](rtc::ArrayView<const uint8_t> data,
+                      const PacketOptions& options) {
+    ++sent_packets;
+    RtpPacketReceived packet;
+    packet.Parse(data);
+    timestamp = packet.Timestamp();
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(send_rtp));
+  ProcessNextFrame();
+  ProcessNextFrame();
+  EXPECT_EQ(sent_packets, 1);
+  uint32_t first_timestamp = timestamp;
+  channel_->StopSend();
+  time_controller_.AdvanceTime(TimeDelta::Seconds(10));
+  channel_->StartSend();
+
+  ProcessNextFrame();
+  ProcessNextFrame();
+  EXPECT_EQ(sent_packets, 2);
+  int64_t timestamp_gap_ms =
+      static_cast<int64_t>(timestamp - first_timestamp) * 1000 / kRtpRateHz;
+  EXPECT_EQ(timestamp_gap_ms, 10020);
+}
+
+TEST_F(ChannelSendTest, FrameTransformerGetsCorrectTimestamp) {
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      rtc::make_ref_counted<MockFrameTransformer>();
+  channel_->SetEncoderToPacketizerFrameTransformer(mock_frame_transformer);
+  rtc::scoped_refptr<TransformedFrameCallback> callback;
+  EXPECT_CALL(*mock_frame_transformer, RegisterTransformedFrameCallback)
+      .WillOnce(SaveArg<0>(&callback));
+  EXPECT_CALL(*mock_frame_transformer, UnregisterTransformedFrameCallback);
+
+  absl::optional<uint32_t> sent_timestamp;
+  auto send_rtp = [&](rtc::ArrayView<const uint8_t> data,
+                      const PacketOptions& options) {
+    RtpPacketReceived packet;
+    packet.Parse(data);
+    if (!sent_timestamp) {
+      sent_timestamp = packet.Timestamp();
+    }
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(send_rtp));
+
+  channel_->StartSend();
+  int64_t transformable_frame_timestamp = -1;
+  EXPECT_CALL(*mock_frame_transformer, Transform)
+      .WillOnce([&](std::unique_ptr<TransformableFrameInterface> frame) {
+        transformable_frame_timestamp = frame->GetTimestamp();
+        callback->OnTransformedFrame(std::move(frame));
+      });
+  // Insert two frames which should trigger a new packet.
+  ProcessNextFrame();
+  ProcessNextFrame();
+
+  // Ensure the RTP timestamp on the frame passed to the transformer
+  // includes the RTP offset and matches the actual RTP timestamp on the sent
+  // packet.
+  EXPECT_EQ_WAIT(transformable_frame_timestamp,
+                 0 + channel_->GetRtpRtcp()->StartTimestamp(), 1000);
+  EXPECT_TRUE_WAIT(sent_timestamp, 1000);
+  EXPECT_EQ(*sent_timestamp, transformable_frame_timestamp);
+}
+}  // namespace
+}  // namespace voe
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/conversion.h b/third_party/libwebrtc/audio/conversion.h
new file mode 100644
index 0000000000..dd71942f6a
--- /dev/null
+++ b/third_party/libwebrtc/audio/conversion.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_CONVERSION_H_
+#define AUDIO_CONVERSION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// Convert fixed point number with 8 bit fractional part, to floating point.
+inline float Q8ToFloat(uint32_t v) {
+  return static_cast<float>(v) / (1 << 8);
+}
+
+// Convert fixed point number with 14 bit fractional part, to floating point.
+inline float Q14ToFloat(uint32_t v) {
+  return static_cast<float>(v) / (1 << 14);
+}
+}  // namespace webrtc
+
+#endif  // AUDIO_CONVERSION_H_
diff --git a/third_party/libwebrtc/audio/mock_voe_channel_proxy.h b/third_party/libwebrtc/audio/mock_voe_channel_proxy.h
new file mode 100644
index 0000000000..29005173df
--- /dev/null
+++ b/third_party/libwebrtc/audio/mock_voe_channel_proxy.h
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
+#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/test/mock_frame_encryptor.h"
+#include "audio/channel_receive.h"
+#include "audio/channel_send.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+class MockChannelReceive : public voe::ChannelReceiveInterface {
+ public:
+  MOCK_METHOD(void, SetNACKStatus, (bool enable, int max_packets), (override));
+  MOCK_METHOD(void, SetNonSenderRttMeasurement, (bool enabled), (override));
+  MOCK_METHOD(void,
+              RegisterReceiverCongestionControlObjects,
+              (PacketRouter*),
+              (override));
+  MOCK_METHOD(void, ResetReceiverCongestionControlObjects, (), (override));
+  MOCK_METHOD(CallReceiveStatistics, GetRTCPStatistics, (), (const, override));
+  MOCK_METHOD(NetworkStatistics,
+              GetNetworkStatistics,
+              (bool),
+              (const, override));
+  MOCK_METHOD(AudioDecodingCallStats,
+              GetDecodingCallStatistics,
+              (),
+              (const, override));
+  MOCK_METHOD(int, GetSpeechOutputLevelFullRange, (), (const, override));
+  MOCK_METHOD(double, GetTotalOutputEnergy, (), (const, override));
+  MOCK_METHOD(double, GetTotalOutputDuration, (), (const, override));
+  MOCK_METHOD(uint32_t, GetDelayEstimate, (), (const, override));
+  MOCK_METHOD(void, SetSink, (AudioSinkInterface*), (override));
+  MOCK_METHOD(void, OnRtpPacket, (const RtpPacketReceived& packet), (override));
+  MOCK_METHOD(void,
+              ReceivedRTCPPacket,
+              (const uint8_t*, size_t length),
+              (override));
+  MOCK_METHOD(void, SetChannelOutputVolumeScaling, (float scaling), (override));
+  MOCK_METHOD(AudioMixer::Source::AudioFrameInfo,
+              GetAudioFrameWithInfo,
+              (int sample_rate_hz, AudioFrame*),
+              (override));
+  MOCK_METHOD(int, PreferredSampleRate, (), (const, override));
+  MOCK_METHOD(void, SetSourceTracker, (SourceTracker*), (override));
+  MOCK_METHOD(void,
+              SetAssociatedSendChannel,
+              (const voe::ChannelSendInterface*),
+              (override));
+  MOCK_METHOD(bool,
+              GetPlayoutRtpTimestamp,
+              (uint32_t*, int64_t*),
+              (const, override));
+  MOCK_METHOD(void,
+              SetEstimatedPlayoutNtpTimestampMs,
+              (int64_t ntp_timestamp_ms, int64_t time_ms),
+              (override));
+  MOCK_METHOD(absl::optional<int64_t>,
+              GetCurrentEstimatedPlayoutNtpTimestampMs,
+              (int64_t now_ms),
+              (const, override));
+  MOCK_METHOD(absl::optional<Syncable::Info>,
+              GetSyncInfo,
+              (),
+              (const, override));
+  MOCK_METHOD(bool, SetMinimumPlayoutDelay, (int delay_ms), (override));
+  MOCK_METHOD(bool, SetBaseMinimumPlayoutDelayMs, (int delay_ms), (override));
+  MOCK_METHOD(int, GetBaseMinimumPlayoutDelayMs, (), (const, override));
+  MOCK_METHOD((absl::optional<std::pair<int, SdpAudioFormat>>),
+              GetReceiveCodec,
+              (),
+              (const, override));
+  MOCK_METHOD(void,
+              SetReceiveCodecs,
+              ((const std::map<int, SdpAudioFormat>& codecs)),
+              (override));
+  MOCK_METHOD(void, StartPlayout, (), (override));
+  MOCK_METHOD(void, StopPlayout, (), (override));
+  MOCK_METHOD(
+      void,
+      SetDepacketizerToDecoderFrameTransformer,
+      (rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
+      (override));
+  MOCK_METHOD(
+      void,
+      SetFrameDecryptor,
+      (rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor),
+      (override));
+  MOCK_METHOD(void, OnLocalSsrcChange, (uint32_t local_ssrc), (override));
+  MOCK_METHOD(uint32_t, GetLocalSsrc, (), (const, override));
+};
+
+class MockChannelSend : public voe::ChannelSendInterface {
+ public:
+  MOCK_METHOD(void,
+              SetEncoder,
+              (int payload_type, std::unique_ptr<AudioEncoder> encoder),
+              (override));
+  MOCK_METHOD(
+      void,
+      ModifyEncoder,
+      (rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier),
+      (override));
+  MOCK_METHOD(void,
+              CallEncoder,
+              (rtc::FunctionView<void(AudioEncoder*)> modifier),
+              (override));
+  MOCK_METHOD(void, SetRTCP_CNAME, (absl::string_view c_name), (override));
+  MOCK_METHOD(void,
+              SetSendAudioLevelIndicationStatus,
+              (bool enable, int id),
+              (override));
+  MOCK_METHOD(void,
+              RegisterSenderCongestionControlObjects,
+              (RtpTransportControllerSendInterface*),
+              (override));
+  MOCK_METHOD(void, ResetSenderCongestionControlObjects, (), (override));
+  MOCK_METHOD(CallSendStatistics, GetRTCPStatistics, (), (const, override));
+  MOCK_METHOD(std::vector<ReportBlockData>,
+              GetRemoteRTCPReportBlocks,
+              (),
+              (const, override));
+  MOCK_METHOD(ANAStats, GetANAStatistics, (), (const, override));
+  MOCK_METHOD(void,
+              RegisterCngPayloadType,
+              (int payload_type, int payload_frequency),
+              (override));
+  MOCK_METHOD(void,
+              SetSendTelephoneEventPayloadType,
+              (int payload_type, int payload_frequency),
+              (override));
+  MOCK_METHOD(bool,
+              SendTelephoneEventOutband,
+              (int event, int duration_ms),
+              (override));
+  MOCK_METHOD(void,
+              OnBitrateAllocation,
+              (BitrateAllocationUpdate update),
+              (override));
+  MOCK_METHOD(void, SetInputMute, (bool muted), (override));
+  MOCK_METHOD(void,
+              ReceivedRTCPPacket,
+              (const uint8_t*, size_t length),
+              (override));
+  MOCK_METHOD(void,
+              ProcessAndEncodeAudio,
+              (std::unique_ptr<AudioFrame>),
+              (override));
+  MOCK_METHOD(RtpRtcpInterface*, GetRtpRtcp, (), (const, override));
+  MOCK_METHOD(int, GetTargetBitrate, (), (const, override));
+  MOCK_METHOD(int64_t, GetRTT, (), (const, override));
+  MOCK_METHOD(void, StartSend, (), (override));
+  MOCK_METHOD(void, StopSend, (), (override));
+  MOCK_METHOD(void,
+              SetFrameEncryptor,
+              (rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor),
+              (override));
+  MOCK_METHOD(
+      void,
+      SetEncoderToPacketizerFrameTransformer,
+      (rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer),
+      (override));
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
diff --git a/third_party/libwebrtc/audio/remix_resample.cc b/third_party/libwebrtc/audio/remix_resample.cc
new file mode 100644
index 0000000000..178af622a1
--- /dev/null
+++ b/third_party/libwebrtc/audio/remix_resample.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/remix_resample.h"
+
+#include "api/audio/audio_frame.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace voe {
+
+void RemixAndResample(const AudioFrame& src_frame,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame) {
+  RemixAndResample(src_frame.data(), src_frame.samples_per_channel_,
+                   src_frame.num_channels_, src_frame.sample_rate_hz_,
+                   resampler, dst_frame);
+  dst_frame->timestamp_ = src_frame.timestamp_;
+  dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
+  dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
+  dst_frame->packet_infos_ = src_frame.packet_infos_;
+}
+
+void RemixAndResample(const int16_t* src_data,
+                      size_t samples_per_channel,
+                      size_t num_channels,
+                      int sample_rate_hz,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame) {
+  const int16_t* audio_ptr = src_data;
+  size_t audio_ptr_num_channels = num_channels;
+  int16_t downmixed_audio[AudioFrame::kMaxDataSizeSamples];
+
+  // Downmix before resampling.
+  if (num_channels > dst_frame->num_channels_) {
+    RTC_DCHECK(num_channels == 2 || num_channels == 4)
+        << "num_channels: " << num_channels;
+    RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2)
+        << "dst_frame->num_channels_: " << dst_frame->num_channels_;
+
+    AudioFrameOperations::DownmixChannels(
+        src_data, num_channels, samples_per_channel, dst_frame->num_channels_,
+        downmixed_audio);
+    audio_ptr = downmixed_audio;
+    audio_ptr_num_channels = dst_frame->num_channels_;
+  }
+
+  if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
+                                    audio_ptr_num_channels) == -1) {
+    RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = "
+                << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
+                << dst_frame->sample_rate_hz_
+                << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
+  }
+
+  // TODO(yujo): for muted input frames, don't resample. Either 1) allow
+  // resampler to return output length without doing the resample, so we know
+  // how much to zero here; or 2) make resampler accept a hint that the input is
+  // zeroed.
+  const size_t src_length = samples_per_channel * audio_ptr_num_channels;
+  int out_length =
+      resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
+                          AudioFrame::kMaxDataSizeSamples);
+  if (out_length == -1) {
+    RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
+                << ", src_length = " << src_length
+                << ", dst_frame->mutable_data() = "
+                << dst_frame->mutable_data();
+  }
+  dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
+
+  // Upmix after resampling.
+  if (num_channels == 1 && dst_frame->num_channels_ == 2) {
+    // The audio in dst_frame really is mono at this point; MonoToStereo will
+    // set this back to stereo.
+    dst_frame->num_channels_ = 1;
+    AudioFrameOperations::UpmixChannels(2, dst_frame);
+  }
+}
+
+}  // namespace voe
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/remix_resample.h b/third_party/libwebrtc/audio/remix_resample.h
new file mode 100644
index 0000000000..bd8da76c6a
--- /dev/null
+++ b/third_party/libwebrtc/audio/remix_resample.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_REMIX_RESAMPLE_H_
+#define AUDIO_REMIX_RESAMPLE_H_
+
+#include "api/audio/audio_frame.h"
+#include "common_audio/resampler/include/push_resampler.h"
+
+namespace webrtc {
+namespace voe {
+
+// Upmix or downmix and resample the audio to `dst_frame`. Expects `dst_frame`
+// to have its sample rate and channels members set to the desired values.
+// Updates the `samples_per_channel_` member accordingly.
+//
+// This version has an AudioFrame `src_frame` as input and sets the output
+// `timestamp_`, `elapsed_time_ms_` and `ntp_time_ms_` members equals to the
+// input ones.
+void RemixAndResample(const AudioFrame& src_frame,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame);
+
+// This version has a pointer to the samples `src_data` as input and receives
+// `samples_per_channel`, `num_channels` and `sample_rate_hz` of the data as
+// parameters.
+void RemixAndResample(const int16_t* src_data,
+                      size_t samples_per_channel,
+                      size_t num_channels,
+                      int sample_rate_hz,
+                      PushResampler<int16_t>* resampler,
+                      AudioFrame* dst_frame);
+
+}  // namespace voe
+}  // namespace webrtc
+
+#endif  // AUDIO_REMIX_RESAMPLE_H_
diff --git a/third_party/libwebrtc/audio/remix_resample_unittest.cc b/third_party/libwebrtc/audio/remix_resample_unittest.cc
new file mode 100644
index 0000000000..31dcfac1fe
--- /dev/null
+++ b/third_party/libwebrtc/audio/remix_resample_unittest.cc
@@ -0,0 +1,276 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/remix_resample.h"
+
+#include <cmath>
+
+#include "common_audio/resampler/include/push_resampler.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace voe {
+namespace {
+
+int GetFrameSize(int sample_rate_hz) {
+  return sample_rate_hz / 100;
+}
+
+class UtilityTest : public ::testing::Test {
+ protected:
+  UtilityTest() {
+    src_frame_.sample_rate_hz_ = 16000;
+    src_frame_.samples_per_channel_ = src_frame_.sample_rate_hz_ / 100;
+    src_frame_.num_channels_ = 1;
+    dst_frame_.CopyFrom(src_frame_);
+    golden_frame_.CopyFrom(src_frame_);
+  }
+
+  void RunResampleTest(int src_channels,
+                       int src_sample_rate_hz,
+                       int dst_channels,
+                       int dst_sample_rate_hz);
+
+  PushResampler<int16_t> resampler_;
+  AudioFrame src_frame_;
+  AudioFrame dst_frame_;
+  AudioFrame golden_frame_;
+};
+
+// Sets the signal value to increase by `data` with every sample. Floats are
+// used so non-integer values result in rounding error, but not an accumulating
+// error.
+void SetMonoFrame(float data, int sample_rate_hz, AudioFrame* frame) {
+  frame->Mute();
+  frame->num_channels_ = 1;
+  frame->sample_rate_hz_ = sample_rate_hz;
+  frame->samples_per_channel_ = GetFrameSize(sample_rate_hz);
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
+    frame_data[i] = static_cast<int16_t>(data * i);
+  }
+}
+
+// Keep the existing sample rate.
+void SetMonoFrame(float data, AudioFrame* frame) {
+  SetMonoFrame(data, frame->sample_rate_hz_, frame);
+}
+
+// Sets the signal value to increase by `left` and `right` with every sample in
+// each channel respectively.
+void SetStereoFrame(float left,
+                    float right,
+                    int sample_rate_hz,
+                    AudioFrame* frame) {
+  frame->Mute();
+  frame->num_channels_ = 2;
+  frame->sample_rate_hz_ = sample_rate_hz;
+  frame->samples_per_channel_ = GetFrameSize(sample_rate_hz);
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
+    frame_data[i * 2] = static_cast<int16_t>(left * i);
+    frame_data[i * 2 + 1] = static_cast<int16_t>(right * i);
+  }
+}
+
+// Keep the existing sample rate.
+void SetStereoFrame(float left, float right, AudioFrame* frame) {
+  SetStereoFrame(left, right, frame->sample_rate_hz_, frame);
+}
+
+// Sets the signal value to increase by `ch1`, `ch2`, `ch3`, `ch4` with every
+// sample in each channel respectively.
+void SetQuadFrame(float ch1,
+                  float ch2,
+                  float ch3,
+                  float ch4,
+                  int sample_rate_hz,
+                  AudioFrame* frame) {
+  frame->Mute();
+  frame->num_channels_ = 4;
+  frame->sample_rate_hz_ = sample_rate_hz;
+  frame->samples_per_channel_ = GetFrameSize(sample_rate_hz);
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
+    frame_data[i * 4] = static_cast<int16_t>(ch1 * i);
+    frame_data[i * 4 + 1] = static_cast<int16_t>(ch2 * i);
+    frame_data[i * 4 + 2] = static_cast<int16_t>(ch3 * i);
+    frame_data[i * 4 + 3] = static_cast<int16_t>(ch4 * i);
+  }
+}
+
+void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
+  EXPECT_EQ(ref_frame.num_channels_, test_frame.num_channels_);
+  EXPECT_EQ(ref_frame.samples_per_channel_, test_frame.samples_per_channel_);
+  EXPECT_EQ(ref_frame.sample_rate_hz_, test_frame.sample_rate_hz_);
+}
+
+// Computes the best SNR based on the error between `ref_frame` and
+// `test_frame`. It allows for up to a `max_delay` in samples between the
+// signals to compensate for the resampling delay.
+float ComputeSNR(const AudioFrame& ref_frame,
+                 const AudioFrame& test_frame,
+                 size_t max_delay) {
+  VerifyParams(ref_frame, test_frame);
+  float best_snr = 0;
+  size_t best_delay = 0;
+  for (size_t delay = 0; delay <= max_delay; delay++) {
+    float mse = 0;
+    float variance = 0;
+    const int16_t* ref_frame_data = ref_frame.data();
+    const int16_t* test_frame_data = test_frame.data();
+    for (size_t i = 0;
+         i < ref_frame.samples_per_channel_ * ref_frame.num_channels_ - delay;
+         i++) {
+      int error = ref_frame_data[i] - test_frame_data[i + delay];
+      mse += error * error;
+      variance += ref_frame_data[i] * ref_frame_data[i];
+    }
+    float snr = 100;  // We assign 100 dB to the zero-error case.
+    if (mse > 0)
+      snr = 10 * std::log10(variance / mse);
+    if (snr > best_snr) {
+      best_snr = snr;
+      best_delay = delay;
+    }
+  }
+  printf("SNR=%.1f dB at delay=%zu\n", best_snr, best_delay);
+  return best_snr;
+}
+
+void VerifyFramesAreEqual(const AudioFrame& ref_frame,
+                          const AudioFrame& test_frame) {
+  VerifyParams(ref_frame, test_frame);
+  const int16_t* ref_frame_data = ref_frame.data();
+  const int16_t* test_frame_data = test_frame.data();
+  for (size_t i = 0;
+       i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) {
+    EXPECT_EQ(ref_frame_data[i], test_frame_data[i]);
+  }
+}
+
+void UtilityTest::RunResampleTest(int src_channels,
+                                  int src_sample_rate_hz,
+                                  int dst_channels,
+                                  int dst_sample_rate_hz) {
+  PushResampler<int16_t> resampler;  // Create a new one with every test.
+  const int16_t kSrcCh1 = 30;  // Shouldn't overflow for any used sample rate.
+  const int16_t kSrcCh2 = 15;
+  const int16_t kSrcCh3 = 22;
+  const int16_t kSrcCh4 = 8;
+  const float resampling_factor =
+      (1.0 * src_sample_rate_hz) / dst_sample_rate_hz;
+  const float dst_ch1 = resampling_factor * kSrcCh1;
+  const float dst_ch2 = resampling_factor * kSrcCh2;
+  const float dst_ch3 = resampling_factor * kSrcCh3;
+  const float dst_ch4 = resampling_factor * kSrcCh4;
+  const float dst_stereo_to_mono = (dst_ch1 + dst_ch2) / 2;
+  const float dst_quad_to_mono = (dst_ch1 + dst_ch2 + dst_ch3 + dst_ch4) / 4;
+  const float dst_quad_to_stereo_ch1 = (dst_ch1 + dst_ch2) / 2;
+  const float dst_quad_to_stereo_ch2 = (dst_ch3 + dst_ch4) / 2;
+  if (src_channels == 1)
+    SetMonoFrame(kSrcCh1, src_sample_rate_hz, &src_frame_);
+  else if (src_channels == 2)
+    SetStereoFrame(kSrcCh1, kSrcCh2, src_sample_rate_hz, &src_frame_);
+  else
+    SetQuadFrame(kSrcCh1, kSrcCh2, kSrcCh3, kSrcCh4, src_sample_rate_hz,
+                 &src_frame_);
+
+  if (dst_channels == 1) {
+    SetMonoFrame(0, dst_sample_rate_hz, &dst_frame_);
+    if (src_channels == 1)
+      SetMonoFrame(dst_ch1, dst_sample_rate_hz, &golden_frame_);
+    else if (src_channels == 2)
+      SetMonoFrame(dst_stereo_to_mono, dst_sample_rate_hz, &golden_frame_);
+    else
+      SetMonoFrame(dst_quad_to_mono, dst_sample_rate_hz, &golden_frame_);
+  } else {
+    SetStereoFrame(0, 0, dst_sample_rate_hz, &dst_frame_);
+    if (src_channels == 1)
+      SetStereoFrame(dst_ch1, dst_ch1, dst_sample_rate_hz, &golden_frame_);
+    else if (src_channels == 2)
+      SetStereoFrame(dst_ch1, dst_ch2, dst_sample_rate_hz, &golden_frame_);
+    else
+      SetStereoFrame(dst_quad_to_stereo_ch1, dst_quad_to_stereo_ch2,
+                     dst_sample_rate_hz, &golden_frame_);
+  }
+
+  // The sinc resampler has a known delay, which we compute here. Multiplying by
+  // two gives us a crude maximum for any resampling, as the old resampler
+  // typically (but not always) has lower delay.
+  static const size_t kInputKernelDelaySamples = 16;
+  const size_t max_delay = static_cast<size_t>(
+      static_cast<double>(dst_sample_rate_hz) / src_sample_rate_hz *
+      kInputKernelDelaySamples * dst_channels * 2);
+  printf("(%d, %d Hz) -> (%d, %d Hz) ",  // SNR reported on the same line later.
+         src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
+  RemixAndResample(src_frame_, &resampler, &dst_frame_);
+
+  if (src_sample_rate_hz == 96000 && dst_sample_rate_hz <= 11025) {
+    // The sinc resampler gives poor SNR at this extreme conversion, but we
+    // expect to see this rarely in practice.
+    EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 14.0f);
+  } else {
+    EXPECT_GT(ComputeSNR(golden_frame_, dst_frame_, max_delay), 46.0f);
+  }
+}
+
+TEST_F(UtilityTest, RemixAndResampleCopyFrameSucceeds) {
+  // Stereo -> stereo.
+  SetStereoFrame(10, 10, &src_frame_);
+  SetStereoFrame(0, 0, &dst_frame_);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(src_frame_, dst_frame_);
+
+  // Mono -> mono.
+  SetMonoFrame(20, &src_frame_);
+  SetMonoFrame(0, &dst_frame_);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(src_frame_, dst_frame_);
+}
+
+TEST_F(UtilityTest, RemixAndResampleMixingOnlySucceeds) {
+  // Stereo -> mono.
+  SetStereoFrame(0, 0, &dst_frame_);
+  SetMonoFrame(10, &src_frame_);
+  SetStereoFrame(10, 10, &golden_frame_);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(dst_frame_, golden_frame_);
+
+  // Mono -> stereo.
+  SetMonoFrame(0, &dst_frame_);
+  SetStereoFrame(10, 20, &src_frame_);
+  SetMonoFrame(15, &golden_frame_);
+  RemixAndResample(src_frame_, &resampler_, &dst_frame_);
+  VerifyFramesAreEqual(golden_frame_, dst_frame_);
+}
+
+TEST_F(UtilityTest, RemixAndResampleSucceeds) {
+  const int kSampleRates[] = {8000,  11025, 16000, 22050,
+                              32000, 44100, 48000, 96000};
+  const int kSrcChannels[] = {1, 2, 4};
+  const int kDstChannels[] = {1, 2};
+
+  for (int src_rate : kSampleRates) {
+    for (int dst_rate : kSampleRates) {
+      for (size_t src_channels : kSrcChannels) {
+        for (size_t dst_channels : kDstChannels) {
+          RunResampleTest(src_channels, src_rate, dst_channels, dst_rate);
+        }
+      }
+    }
+  }
+}
+
+}  // namespace
+}  // namespace voe
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc b/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc
new file mode 100644
index 0000000000..b1e2712f60
--- /dev/null
+++ b/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/test/audio_end_to_end_test.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "api/task_queue/task_queue_base.h"
+#include "call/fake_network_pipe.h"
+#include "call/simulated_network.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+#include "test/video_test_constants.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr int kSampleRate = 48000;
+
+}  // namespace
+
+AudioEndToEndTest::AudioEndToEndTest()
+    : EndToEndTest(VideoTestConstants::kDefaultTimeout) {}
+
+size_t AudioEndToEndTest::GetNumVideoStreams() const {
+  return 0;
+}
+
+size_t AudioEndToEndTest::GetNumAudioStreams() const {
+  return 1;
+}
+
+size_t AudioEndToEndTest::GetNumFlexfecStreams() const {
+  return 0;
+}
+
+std::unique_ptr<TestAudioDeviceModule::Capturer>
+AudioEndToEndTest::CreateCapturer() {
+  return TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, kSampleRate);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer>
+AudioEndToEndTest::CreateRenderer() {
+  return TestAudioDeviceModule::CreateDiscardRenderer(kSampleRate);
+}
+
+void AudioEndToEndTest::OnFakeAudioDevicesCreated(
+    AudioDeviceModule* send_audio_device,
+    AudioDeviceModule* recv_audio_device) {
+  send_audio_device_ = send_audio_device;
+}
+
+void AudioEndToEndTest::ModifyAudioConfigs(
+    AudioSendStream::Config* send_config,
+    std::vector<AudioReceiveStreamInterface::Config>* receive_configs) {
+  // Large bitrate by default.
+  const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2,
+                                              {{"stereo", "1"}});
+  send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
+      test::VideoTestConstants::kAudioSendPayloadType, kDefaultFormat);
+  send_config->min_bitrate_bps = 32000;
+  send_config->max_bitrate_bps = 32000;
+}
+
+void AudioEndToEndTest::OnAudioStreamsCreated(
+    AudioSendStream* send_stream,
+    const std::vector<AudioReceiveStreamInterface*>& receive_streams) {
+  ASSERT_NE(nullptr, send_stream);
+  ASSERT_EQ(1u, receive_streams.size());
+  ASSERT_NE(nullptr, receive_streams[0]);
+  send_stream_ = send_stream;
+  receive_stream_ = receive_streams[0];
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/test/audio_end_to_end_test.h b/third_party/libwebrtc/audio/test/audio_end_to_end_test.h
new file mode 100644
index 0000000000..d326b790ff
--- /dev/null
+++ b/third_party/libwebrtc/audio/test/audio_end_to_end_test.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
+#define AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/task_queue/task_queue_base.h"
+#include "api/test/simulated_network.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "test/call_test.h"
+
+namespace webrtc {
+namespace test {
+
+class AudioEndToEndTest : public test::EndToEndTest {
+ public:
+  AudioEndToEndTest();
+
+ protected:
+  AudioDeviceModule* send_audio_device() { return send_audio_device_; }
+  const AudioSendStream* send_stream() const { return send_stream_; }
+  const AudioReceiveStreamInterface* receive_stream() const {
+    return receive_stream_;
+  }
+
+  size_t GetNumVideoStreams() const override;
+  size_t GetNumAudioStreams() const override;
+  size_t GetNumFlexfecStreams() const override;
+
+  std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override;
+  std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override;
+
+  void OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device,
+                                 AudioDeviceModule* recv_audio_device) override;
+
+  void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+                          std::vector<AudioReceiveStreamInterface::Config>*
+                              receive_configs) override;
+  void OnAudioStreamsCreated(AudioSendStream* send_stream,
+                             const std::vector<AudioReceiveStreamInterface*>&
+                                 receive_streams) override;
+
+ private:
+  AudioDeviceModule* send_audio_device_ = nullptr;
+  AudioSendStream* send_stream_ = nullptr;
+  AudioReceiveStreamInterface* receive_stream_ = nullptr;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
diff --git a/third_party/libwebrtc/audio/test/audio_stats_test.cc b/third_party/libwebrtc/audio/test/audio_stats_test.cc
new file mode 100644
index 0000000000..e8521cfe99
--- /dev/null
+++ b/third_party/libwebrtc/audio/test/audio_stats_test.cc
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/test/audio_end_to_end_test.h"
+#include "rtc_base/numerics/safe_compare.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+// Wait half a second between stopping sending and stopping receiving audio.
+constexpr int kExtraRecordTimeMs = 500;
+
+bool IsNear(int reference, int v) {
+  // Margin is 10%.
+  const int error = reference / 10 + 1;
+  return std::abs(reference - v) <= error;
+}
+
+class NoLossTest : public AudioEndToEndTest {
+ public:
+  const int kTestDurationMs = 8000;
+  const int kBytesSent = 69351;
+  const int32_t kPacketsSent = 400;
+  const int64_t kRttMs = 100;
+
+  NoLossTest() = default;
+
+  BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
+    BuiltInNetworkBehaviorConfig pipe_config;
+    pipe_config.queue_delay_ms = kRttMs / 2;
+    return pipe_config;
+  }
+
+  void PerformTest() override {
+    SleepMs(kTestDurationMs);
+    send_audio_device()->StopRecording();
+    // and some extra time to account for network delay.
+    SleepMs(GetSendTransportConfig().queue_delay_ms + kExtraRecordTimeMs);
+  }
+
+  void OnStreamsStopped() override {
+    AudioSendStream::Stats send_stats = send_stream()->GetStats();
+    EXPECT_PRED2(IsNear, kBytesSent, send_stats.payload_bytes_sent);
+    EXPECT_PRED2(IsNear, kPacketsSent, send_stats.packets_sent);
+    EXPECT_EQ(0, send_stats.packets_lost);
+    EXPECT_EQ(0.0f, send_stats.fraction_lost);
+    EXPECT_EQ("opus", send_stats.codec_name);
+    // send_stats.jitter_ms
+    EXPECT_PRED2(IsNear, kRttMs, send_stats.rtt_ms);
+    // Send level is 0 because it is cleared in TransmitMixer::StopSend().
+    EXPECT_EQ(0, send_stats.audio_level);
+    // send_stats.total_input_energy
+    // send_stats.total_input_duration
+    EXPECT_FALSE(send_stats.apm_statistics.delay_median_ms);
+    EXPECT_FALSE(send_stats.apm_statistics.delay_standard_deviation_ms);
+    EXPECT_FALSE(send_stats.apm_statistics.echo_return_loss);
+    EXPECT_FALSE(send_stats.apm_statistics.echo_return_loss_enhancement);
+    EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood);
+    EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood_recent_max);
+
+    AudioReceiveStreamInterface::Stats recv_stats =
+        receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true);
+    EXPECT_PRED2(IsNear, kBytesSent, recv_stats.payload_bytes_received);
+    EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_received);
+    EXPECT_EQ(0, recv_stats.packets_lost);
+    EXPECT_EQ("opus", send_stats.codec_name);
+    // recv_stats.jitter_ms
+    // recv_stats.jitter_buffer_ms
+    EXPECT_EQ(20u, recv_stats.jitter_buffer_preferred_ms);
+    // recv_stats.delay_estimate_ms
+    // Receive level is 0 because it is cleared in Channel::StopPlayout().
+    EXPECT_EQ(0, recv_stats.audio_level);
+    // recv_stats.total_output_energy
+    // recv_stats.total_samples_received
+    // recv_stats.total_output_duration
+    // recv_stats.concealed_samples
+    // recv_stats.expand_rate
+    // recv_stats.speech_expand_rate
+    EXPECT_EQ(0.0, recv_stats.secondary_decoded_rate);
+    EXPECT_EQ(0.0, recv_stats.secondary_discarded_rate);
+    EXPECT_EQ(0.0, recv_stats.accelerate_rate);
+    EXPECT_EQ(0.0, recv_stats.preemptive_expand_rate);
+    EXPECT_EQ(0, recv_stats.decoding_calls_to_silence_generator);
+    // recv_stats.decoding_calls_to_neteq
+    // recv_stats.decoding_normal
+    // recv_stats.decoding_plc
+    EXPECT_EQ(0, recv_stats.decoding_cng);
+    // recv_stats.decoding_plc_cng
+    // recv_stats.decoding_muted_output
+    // Capture start time is -1 because we do not have an associated send stream
+    // on the receiver side.
+    EXPECT_EQ(-1, recv_stats.capture_start_ntp_time_ms);
+
+    // Match these stats between caller and receiver.
+    EXPECT_EQ(send_stats.local_ssrc, recv_stats.remote_ssrc);
+    EXPECT_EQ(*send_stats.codec_payload_type, *recv_stats.codec_payload_type);
+  }
+};
+}  // namespace
+
+using AudioStatsTest = CallTest;
+
+TEST_F(AudioStatsTest, DISABLED_NoLoss) {
+  NoLossTest test;
+  RunBaseTest(&test);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/test/nack_test.cc b/third_party/libwebrtc/audio/test/nack_test.cc
new file mode 100644
index 0000000000..b36adf8991
--- /dev/null
+++ b/third_party/libwebrtc/audio/test/nack_test.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/test/audio_end_to_end_test.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+using NackTest = CallTest;
+
+TEST_F(NackTest, ShouldNackInLossyNetwork) {
+  class NackTest : public AudioEndToEndTest {
+   public:
+    const int kTestDurationMs = 2000;
+    const int64_t kRttMs = 30;
+    const int64_t kLossPercent = 30;
+    const int kNackHistoryMs = 1000;
+
+    BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
+      BuiltInNetworkBehaviorConfig pipe_config;
+      pipe_config.queue_delay_ms = kRttMs / 2;
+      pipe_config.loss_percent = kLossPercent;
+      return pipe_config;
+    }
+
+    void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+                            std::vector<AudioReceiveStreamInterface::Config>*
+                                receive_configs) override {
+      ASSERT_EQ(receive_configs->size(), 1U);
+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackHistoryMs;
+      AudioEndToEndTest::ModifyAudioConfigs(send_config, receive_configs);
+    }
+
+    void PerformTest() override { SleepMs(kTestDurationMs); }
+
+    void OnStreamsStopped() override {
+      AudioReceiveStreamInterface::Stats recv_stats =
+          receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true);
+      EXPECT_GT(recv_stats.nacks_sent, 0U);
+      AudioSendStream::Stats send_stats = send_stream()->GetStats();
+      EXPECT_GT(send_stats.retransmitted_packets_sent, 0U);
+      EXPECT_GT(send_stats.nacks_received, 0U);
+    }
+  } test;
+
+  RunBaseTest(&test);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/test/non_sender_rtt_test.cc b/third_party/libwebrtc/audio/test/non_sender_rtt_test.cc
new file mode 100644
index 0000000000..278193e335
--- /dev/null
+++ b/third_party/libwebrtc/audio/test/non_sender_rtt_test.cc
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/test/audio_end_to_end_test.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+using NonSenderRttTest = CallTest;
+
+TEST_F(NonSenderRttTest, NonSenderRttStats) {
+  class NonSenderRttTest : public AudioEndToEndTest {
+   public:
+    const int kLongTimeoutMs = 20000;
+    const int64_t kRttMs = 30;
+
+    explicit NonSenderRttTest(TaskQueueBase* task_queue)
+        : task_queue_(task_queue) {}
+
+    BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
+      BuiltInNetworkBehaviorConfig pipe_config;
+      pipe_config.queue_delay_ms = kRttMs / 2;
+      return pipe_config;
+    }
+
+    void ModifyAudioConfigs(AudioSendStream::Config* send_config,
+                            std::vector<AudioReceiveStreamInterface::Config>*
+                                receive_configs) override {
+      ASSERT_EQ(receive_configs->size(), 1U);
+      (*receive_configs)[0].enable_non_sender_rtt = true;
+      AudioEndToEndTest::ModifyAudioConfigs(send_config, receive_configs);
+      send_config->send_codec_spec->enable_non_sender_rtt = true;
+    }
+
+    void PerformTest() override {
+      // Wait until we have an RTT measurement, but no longer than
+      // `kLongTimeoutMs`. This usually takes around 5 seconds, but in rare
+      // cases it can take more than 10 seconds.
+      EXPECT_TRUE_WAIT(HasRoundTripTimeMeasurement(), kLongTimeoutMs);
+    }
+
+    void OnStreamsStopped() override {
+      AudioReceiveStreamInterface::Stats recv_stats =
+          receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true);
+      EXPECT_GT(recv_stats.round_trip_time_measurements, 0);
+      ASSERT_TRUE(recv_stats.round_trip_time.has_value());
+      EXPECT_GT(recv_stats.round_trip_time->ms(), 0);
+      EXPECT_GE(recv_stats.total_round_trip_time.ms(),
+                recv_stats.round_trip_time->ms());
+    }
+
+   protected:
+    bool HasRoundTripTimeMeasurement() {
+      bool has_rtt = false;
+      // GetStats() can only be called on `task_queue_`, block while we check.
+      SendTask(task_queue_, [this, &has_rtt]() {
+        if (receive_stream() &&
+            receive_stream()->GetStats(true).round_trip_time_measurements > 0) {
+          has_rtt = true;
+        }
+      });
+      return has_rtt;
+    }
+
+   private:
+    TaskQueueBase* task_queue_;
+  } test(task_queue());
+
+  RunBaseTest(&test);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/BUILD.gn b/third_party/libwebrtc/audio/utility/BUILD.gn
new file mode 100644
index 0000000000..983b6286e4
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/BUILD.gn
@@ -0,0 +1,56 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+import("../../webrtc.gni")
+
+group("utility") {
+  deps = [ ":audio_frame_operations" ]
+}
+
+rtc_library("audio_frame_operations") {
+  visibility = [ "*" ]
+  sources = [
+    "audio_frame_operations.cc",
+    "audio_frame_operations.h",
+    "channel_mixer.cc",
+    "channel_mixer.h",
+    "channel_mixing_matrix.cc",
+    "channel_mixing_matrix.h",
+  ]
+
+  deps = [
+    "../../api/audio:audio_frame_api",
+    "../../common_audio",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:safe_conversions",
+    "../../system_wrappers:field_trial",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("utility_tests") {
+    testonly = true
+    sources = [
+      "audio_frame_operations_unittest.cc",
+      "channel_mixer_unittest.cc",
+      "channel_mixing_matrix_unittest.cc",
+    ]
+    deps = [
+      ":audio_frame_operations",
+      "../../api/audio:audio_frame_api",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:stringutils",
+      "../../test:field_trial",
+      "../../test:test_support",
+      "//testing/gtest",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations.cc b/third_party/libwebrtc/audio/utility/audio_frame_operations.cc
new file mode 100644
index 0000000000..1b936c239b
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/audio_frame_operations.cc
@@ -0,0 +1,294 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/audio_frame_operations.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <utility>
+
+#include "common_audio/include/audio_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace {
+
+// 2.7ms @ 48kHz, 4ms @ 32kHz, 8ms @ 16kHz.
+const size_t kMuteFadeFrames = 128;
+const float kMuteFadeInc = 1.0f / kMuteFadeFrames;
+
+}  // namespace
+
+void AudioFrameOperations::Add(const AudioFrame& frame_to_add,
+                               AudioFrame* result_frame) {
+  // Sanity check.
+  RTC_DCHECK(result_frame);
+  RTC_DCHECK_GT(result_frame->num_channels_, 0);
+  RTC_DCHECK_EQ(result_frame->num_channels_, frame_to_add.num_channels_);
+
+  bool no_previous_data = result_frame->muted();
+  if (result_frame->samples_per_channel_ != frame_to_add.samples_per_channel_) {
+    // Special case we have no data to start with.
+    RTC_DCHECK_EQ(result_frame->samples_per_channel_, 0);
+    result_frame->samples_per_channel_ = frame_to_add.samples_per_channel_;
+    no_previous_data = true;
+  }
+
+  if (result_frame->vad_activity_ == AudioFrame::kVadActive ||
+      frame_to_add.vad_activity_ == AudioFrame::kVadActive) {
+    result_frame->vad_activity_ = AudioFrame::kVadActive;
+  } else if (result_frame->vad_activity_ == AudioFrame::kVadUnknown ||
+             frame_to_add.vad_activity_ == AudioFrame::kVadUnknown) {
+    result_frame->vad_activity_ = AudioFrame::kVadUnknown;
+  }
+
+  if (result_frame->speech_type_ != frame_to_add.speech_type_)
+    result_frame->speech_type_ = AudioFrame::kUndefined;
+
+  if (!frame_to_add.muted()) {
+    const int16_t* in_data = frame_to_add.data();
+    int16_t* out_data = result_frame->mutable_data();
+    size_t length =
+        frame_to_add.samples_per_channel_ * frame_to_add.num_channels_;
+    if (no_previous_data) {
+      std::copy(in_data, in_data + length, out_data);
+    } else {
+      for (size_t i = 0; i < length; i++) {
+        const int32_t wrap_guard = static_cast<int32_t>(out_data[i]) +
+                                   static_cast<int32_t>(in_data[i]);
+        out_data[i] = rtc::saturated_cast<int16_t>(wrap_guard);
+      }
+    }
+  }
+}
+
+int AudioFrameOperations::MonoToStereo(AudioFrame* frame) {
+  if (frame->num_channels_ != 1) {
+    return -1;
+  }
+  UpmixChannels(2, frame);
+  return 0;
+}
+
+int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
+  if (frame->num_channels_ != 2) {
+    return -1;
+  }
+  DownmixChannels(1, frame);
+  return frame->num_channels_ == 1 ? 0 : -1;
+}
+
+void AudioFrameOperations::QuadToStereo(const int16_t* src_audio,
+                                        size_t samples_per_channel,
+                                        int16_t* dst_audio) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
+    dst_audio[i * 2] =
+        (static_cast<int32_t>(src_audio[4 * i]) + src_audio[4 * i + 1]) >> 1;
+    dst_audio[i * 2 + 1] =
+        (static_cast<int32_t>(src_audio[4 * i + 2]) + src_audio[4 * i + 3]) >>
+        1;
+  }
+}
+
+int AudioFrameOperations::QuadToStereo(AudioFrame* frame) {
+  if (frame->num_channels_ != 4) {
+    return -1;
+  }
+
+  RTC_DCHECK_LE(frame->samples_per_channel_ * 4,
+                AudioFrame::kMaxDataSizeSamples);
+
+  if (!frame->muted()) {
+    QuadToStereo(frame->data(), frame->samples_per_channel_,
+                 frame->mutable_data());
+  }
+  frame->num_channels_ = 2;
+
+  return 0;
+}
+
+void AudioFrameOperations::DownmixChannels(const int16_t* src_audio,
+                                           size_t src_channels,
+                                           size_t samples_per_channel,
+                                           size_t dst_channels,
+                                           int16_t* dst_audio) {
+  if (src_channels > 1 && dst_channels == 1) {
+    DownmixInterleavedToMono(src_audio, samples_per_channel, src_channels,
+                             dst_audio);
+    return;
+  } else if (src_channels == 4 && dst_channels == 2) {
+    QuadToStereo(src_audio, samples_per_channel, dst_audio);
+    return;
+  }
+
+  RTC_DCHECK_NOTREACHED() << "src_channels: " << src_channels
+                          << ", dst_channels: " << dst_channels;
+}
+
+void AudioFrameOperations::DownmixChannels(size_t dst_channels,
+                                           AudioFrame* frame) {
+  RTC_DCHECK_LE(frame->samples_per_channel_ * frame->num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
+  if (frame->num_channels_ > 1 && dst_channels == 1) {
+    if (!frame->muted()) {
+      DownmixInterleavedToMono(frame->data(), frame->samples_per_channel_,
+                               frame->num_channels_, frame->mutable_data());
+    }
+    frame->num_channels_ = 1;
+  } else if (frame->num_channels_ == 4 && dst_channels == 2) {
+    int err = QuadToStereo(frame);
+    RTC_DCHECK_EQ(err, 0);
+  } else {
+    RTC_DCHECK_NOTREACHED() << "src_channels: " << frame->num_channels_
+                            << ", dst_channels: " << dst_channels;
+  }
+}
+
+void AudioFrameOperations::UpmixChannels(size_t target_number_of_channels,
+                                         AudioFrame* frame) {
+  RTC_DCHECK_EQ(frame->num_channels_, 1);
+  RTC_DCHECK_LE(frame->samples_per_channel_ * target_number_of_channels,
+                AudioFrame::kMaxDataSizeSamples);
+
+  if (frame->num_channels_ != 1 ||
+      frame->samples_per_channel_ * target_number_of_channels >
+          AudioFrame::kMaxDataSizeSamples) {
+    return;
+  }
+
+  if (!frame->muted()) {
+    // Up-mixing done in place. Going backwards through the frame ensure nothing
+    // is irrevocably overwritten.
+    int16_t* frame_data = frame->mutable_data();
+    for (int i = frame->samples_per_channel_ - 1; i >= 0; i--) {
+      for (size_t j = 0; j < target_number_of_channels; ++j) {
+        frame_data[target_number_of_channels * i + j] = frame_data[i];
+      }
+    }
+  }
+  frame->num_channels_ = target_number_of_channels;
+}
+
+void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  if (frame->num_channels_ != 2 || frame->muted()) {
+    return;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
+    std::swap(frame_data[i], frame_data[i + 1]);
+  }
+}
+
+void AudioFrameOperations::Mute(AudioFrame* frame,
+                                bool previous_frame_muted,
+                                bool current_frame_muted) {
+  RTC_DCHECK(frame);
+  if (!previous_frame_muted && !current_frame_muted) {
+    // Not muted, don't touch.
+  } else if (previous_frame_muted && current_frame_muted) {
+    // Frame fully muted.
+    size_t total_samples = frame->samples_per_channel_ * frame->num_channels_;
+    RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples, total_samples);
+    frame->Mute();
+  } else {
+    // Fade is a no-op on a muted frame.
+    if (frame->muted()) {
+      return;
+    }
+
+    // Limit number of samples to fade, if frame isn't long enough.
+    size_t count = kMuteFadeFrames;
+    float inc = kMuteFadeInc;
+    if (frame->samples_per_channel_ < kMuteFadeFrames) {
+      count = frame->samples_per_channel_;
+      if (count > 0) {
+        inc = 1.0f / count;
+      }
+    }
+
+    size_t start = 0;
+    size_t end = count;
+    float start_g = 0.0f;
+    if (current_frame_muted) {
+      // Fade out the last `count` samples of frame.
+      RTC_DCHECK(!previous_frame_muted);
+      start = frame->samples_per_channel_ - count;
+      end = frame->samples_per_channel_;
+      start_g = 1.0f;
+      inc = -inc;
+    } else {
+      // Fade in the first `count` samples of frame.
+      RTC_DCHECK(previous_frame_muted);
+    }
+
+    // Perform fade.
+    int16_t* frame_data = frame->mutable_data();
+    size_t channels = frame->num_channels_;
+    for (size_t j = 0; j < channels; ++j) {
+      float g = start_g;
+      for (size_t i = start * channels; i < end * channels; i += channels) {
+        g += inc;
+        frame_data[i + j] *= g;
+      }
+    }
+  }
+}
+
+void AudioFrameOperations::Mute(AudioFrame* frame) {
+  Mute(frame, true, true);
+}
+
+void AudioFrameOperations::ApplyHalfGain(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  RTC_DCHECK_GT(frame->num_channels_, 0);
+  if (frame->num_channels_ < 1 || frame->muted()) {
+    return;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       i++) {
+    frame_data[i] = frame_data[i] >> 1;
+  }
+}
+
+int AudioFrameOperations::Scale(float left, float right, AudioFrame* frame) {
+  if (frame->num_channels_ != 2) {
+    return -1;
+  } else if (frame->muted()) {
+    return 0;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
+    frame_data[2 * i] = static_cast<int16_t>(left * frame_data[2 * i]);
+    frame_data[2 * i + 1] = static_cast<int16_t>(right * frame_data[2 * i + 1]);
+  }
+  return 0;
+}
+
+int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame* frame) {
+  if (frame->muted()) {
+    return 0;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       i++) {
+    frame_data[i] = rtc::saturated_cast<int16_t>(scale * frame_data[i]);
+  }
+  return 0;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations.h b/third_party/libwebrtc/audio/utility/audio_frame_operations.h
new file mode 100644
index 0000000000..2a5f29f4f5
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/audio_frame_operations.h
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
+#define AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "api/audio/audio_frame.h"
+
+namespace webrtc {
+
+// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
+// Change reference parameters to pointers. Consider using a namespace rather
+// than a class.
+class AudioFrameOperations {
+ public:
+  // Add samples in `frame_to_add` with samples in `result_frame`
+  // putting the results in `results_frame`.  The fields
+  // `vad_activity_` and `speech_type_` of the result frame are
+  // updated. If `result_frame` is empty (`samples_per_channel_`==0),
+  // the samples in `frame_to_add` are added to it.  The number of
+  // channels and number of samples per channel must match except when
+  // `result_frame` is empty.
+  static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
+
+  // `frame.num_channels_` will be updated. This version checks for sufficient
+  // buffer size and that `num_channels_` is mono. Use UpmixChannels
+  // instead. TODO(bugs.webrtc.org/8649): remove.
+  ABSL_DEPRECATED("bugs.webrtc.org/8649")
+  static int MonoToStereo(AudioFrame* frame);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` is stereo. Use DownmixChannels
+  // instead. TODO(bugs.webrtc.org/8649): remove.
+  ABSL_DEPRECATED("bugs.webrtc.org/8649")
+  static int StereoToMono(AudioFrame* frame);
+
+  // Downmixes 4 channels `src_audio` to stereo `dst_audio`. This is an in-place
+  // operation, meaning `src_audio` and `dst_audio` may point to the same
+  // buffer.
+  static void QuadToStereo(const int16_t* src_audio,
+                           size_t samples_per_channel,
+                           int16_t* dst_audio);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` is 4 channels.
+  static int QuadToStereo(AudioFrame* frame);
+
+  // Downmixes `src_channels` `src_audio` to `dst_channels` `dst_audio`.
+  // This is an in-place operation, meaning `src_audio` and `dst_audio`
+  // may point to the same buffer. Supported channel combinations are
+  // Stereo to Mono, Quad to Mono, and Quad to Stereo.
+  static void DownmixChannels(const int16_t* src_audio,
+                              size_t src_channels,
+                              size_t samples_per_channel,
+                              size_t dst_channels,
+                              int16_t* dst_audio);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` and `dst_channels` are valid and performs relevant downmix.
+  // Supported channel combinations are N channels to Mono, and Quad to Stereo.
+  static void DownmixChannels(size_t dst_channels, AudioFrame* frame);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` and `dst_channels` are valid and performs relevant
+  // downmix. Supported channel combinations are Mono to N
+  // channels. The single channel is replicated.
+  static void UpmixChannels(size_t target_number_of_channels,
+                            AudioFrame* frame);
+
+  // Swap the left and right channels of `frame`. Fails silently if `frame` is
+  // not stereo.
+  static void SwapStereoChannels(AudioFrame* frame);
+
+  // Conditionally zero out contents of `frame` for implementing audio mute:
+  //  `previous_frame_muted` &&  `current_frame_muted` - Zero out whole frame.
+  //  `previous_frame_muted` && !`current_frame_muted` - Fade-in at frame start.
+  // !`previous_frame_muted` &&  `current_frame_muted` - Fade-out at frame end.
+  // !`previous_frame_muted` && !`current_frame_muted` - Leave frame untouched.
+  static void Mute(AudioFrame* frame,
+                   bool previous_frame_muted,
+                   bool current_frame_muted);
+
+  // Zero out contents of frame.
+  static void Mute(AudioFrame* frame);
+
+  // Halve samples in `frame`.
+  static void ApplyHalfGain(AudioFrame* frame);
+
+  static int Scale(float left, float right, AudioFrame* frame);
+
+  static int ScaleWithSat(float scale, AudioFrame* frame);
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations_gn/moz.build b/third_party/libwebrtc/audio/utility/audio_frame_operations_gn/moz.build
new file mode 100644
index 0000000000..e215792f64
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/audio_frame_operations_gn/moz.build
@@ -0,0 +1,238 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/audio/utility/audio_frame_operations.cc",
+    "/third_party/libwebrtc/audio/utility/channel_mixer.cc",
+    "/third_party/libwebrtc/audio/utility/channel_mixing_matrix.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_frame_operations_gn")
diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations_unittest.cc b/third_party/libwebrtc/audio/utility/audio_frame_operations_unittest.cc
new file mode 100644
index 0000000000..1a2c16e45f
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/audio_frame_operations_unittest.cc
@@ -0,0 +1,622 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/audio_frame_operations.h"
+
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+class AudioFrameOperationsTest : public ::testing::Test {
+ protected:
+  AudioFrameOperationsTest() {
+    // Set typical values.
+    frame_.samples_per_channel_ = 320;
+    frame_.num_channels_ = 2;
+  }
+
+  AudioFrame frame_;
+};
+
+class AudioFrameOperationsDeathTest : public AudioFrameOperationsTest {};
+
+void SetFrameData(int16_t ch1,
+                  int16_t ch2,
+                  int16_t ch3,
+                  int16_t ch4,
+                  AudioFrame* frame) {
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * 4; i += 4) {
+    frame_data[i] = ch1;
+    frame_data[i + 1] = ch2;
+    frame_data[i + 2] = ch3;
+    frame_data[i + 3] = ch4;
+  }
+}
+
+void SetFrameData(int16_t left, int16_t right, AudioFrame* frame) {
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
+    frame_data[i] = left;
+    frame_data[i + 1] = right;
+  }
+}
+
+void SetFrameData(int16_t data, AudioFrame* frame) {
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       i++) {
+    frame_data[i] = data;
+  }
+}
+
+void VerifyFramesAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
+  EXPECT_EQ(frame1.num_channels_, frame2.num_channels_);
+  EXPECT_EQ(frame1.samples_per_channel_, frame2.samples_per_channel_);
+  const int16_t* frame1_data = frame1.data();
+  const int16_t* frame2_data = frame2.data();
+  for (size_t i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_;
+       i++) {
+    EXPECT_EQ(frame1_data[i], frame2_data[i]);
+  }
+  EXPECT_EQ(frame1.muted(), frame2.muted());
+}
+
+void InitFrame(AudioFrame* frame,
+               size_t channels,
+               size_t samples_per_channel,
+               int16_t left_data,
+               int16_t right_data) {
+  RTC_DCHECK(frame);
+  RTC_DCHECK_GE(2, channels);
+  RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples,
+                samples_per_channel * channels);
+  frame->samples_per_channel_ = samples_per_channel;
+  frame->num_channels_ = channels;
+  if (channels == 2) {
+    SetFrameData(left_data, right_data, frame);
+  } else if (channels == 1) {
+    SetFrameData(left_data, frame);
+  }
+}
+
+int16_t GetChannelData(const AudioFrame& frame, size_t channel, size_t index) {
+  RTC_DCHECK_LT(channel, frame.num_channels_);
+  RTC_DCHECK_LT(index, frame.samples_per_channel_);
+  return frame.data()[index * frame.num_channels_ + channel];
+}
+
+void VerifyFrameDataBounds(const AudioFrame& frame,
+                           size_t channel,
+                           int16_t max,
+                           int16_t min) {
+  for (size_t i = 0; i < frame.samples_per_channel_; ++i) {
+    int16_t s = GetChannelData(frame, channel, i);
+    EXPECT_LE(min, s);
+    EXPECT_GE(max, s);
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST_F(AudioFrameOperationsDeathTest, MonoToStereoFailsWithBadParameters) {
+  EXPECT_DEATH(AudioFrameOperations::UpmixChannels(2, &frame_), "");
+  frame_.samples_per_channel_ = AudioFrame::kMaxDataSizeSamples;
+  frame_.num_channels_ = 1;
+  EXPECT_DEATH(AudioFrameOperations::UpmixChannels(2, &frame_), "");
+}
+#endif
+
+TEST_F(AudioFrameOperationsTest, MonoToStereoSucceeds) {
+  frame_.num_channels_ = 1;
+  SetFrameData(1, &frame_);
+
+  AudioFrameOperations::UpmixChannels(2, &frame_);
+  EXPECT_EQ(2u, frame_.num_channels_);
+
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = 320;
+  stereo_frame.num_channels_ = 2;
+  SetFrameData(1, 1, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, MonoToStereoMuted) {
+  frame_.num_channels_ = 1;
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::UpmixChannels(2, &frame_);
+  EXPECT_EQ(2u, frame_.num_channels_);
+  EXPECT_TRUE(frame_.muted());
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST_F(AudioFrameOperationsDeathTest, StereoToMonoFailsWithBadParameters) {
+  frame_.num_channels_ = 1;
+  EXPECT_DEATH(AudioFrameOperations::DownmixChannels(1, &frame_), "");
+}
+#endif
+
+TEST_F(AudioFrameOperationsTest, StereoToMonoSucceeds) {
+  SetFrameData(4, 2, &frame_);
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(3, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, StereoToMonoMuted) {
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, StereoToMonoBufferSucceeds) {
+  AudioFrame target_frame;
+  SetFrameData(4, 2, &frame_);
+
+  target_frame.num_channels_ = 1;
+  target_frame.samples_per_channel_ = frame_.samples_per_channel_;
+
+  AudioFrameOperations::DownmixChannels(frame_.data(), 2,
+                                        frame_.samples_per_channel_, 1,
+                                        target_frame.mutable_data());
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(3, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, target_frame);
+}
+
+TEST_F(AudioFrameOperationsTest, StereoToMonoDoesNotWrapAround) {
+  SetFrameData(-32768, -32768, &frame_);
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(-32768, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToMonoSucceeds) {
+  frame_.num_channels_ = 4;
+  SetFrameData(4, 2, 6, 8, &frame_);
+
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(5, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToMonoMuted) {
+  frame_.num_channels_ = 4;
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToMonoBufferSucceeds) {
+  AudioFrame target_frame;
+  frame_.num_channels_ = 4;
+  SetFrameData(4, 2, 6, 8, &frame_);
+
+  target_frame.num_channels_ = 1;
+  target_frame.samples_per_channel_ = frame_.samples_per_channel_;
+
+  AudioFrameOperations::DownmixChannels(frame_.data(), 4,
+                                        frame_.samples_per_channel_, 1,
+                                        target_frame.mutable_data());
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(5, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, target_frame);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToMonoDoesNotWrapAround) {
+  frame_.num_channels_ = 4;
+  SetFrameData(-32768, -32768, -32768, -32768, &frame_);
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(-32768, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToStereoFailsWithBadParameters) {
+  frame_.num_channels_ = 1;
+  EXPECT_EQ(-1, AudioFrameOperations::QuadToStereo(&frame_));
+  frame_.num_channels_ = 2;
+  EXPECT_EQ(-1, AudioFrameOperations::QuadToStereo(&frame_));
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToStereoSucceeds) {
+  frame_.num_channels_ = 4;
+  SetFrameData(4, 2, 6, 8, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::QuadToStereo(&frame_));
+
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = 320;
+  stereo_frame.num_channels_ = 2;
+  SetFrameData(3, 7, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToStereoMuted) {
+  frame_.num_channels_ = 4;
+  ASSERT_TRUE(frame_.muted());
+  EXPECT_EQ(0, AudioFrameOperations::QuadToStereo(&frame_));
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToStereoBufferSucceeds) {
+  AudioFrame target_frame;
+  frame_.num_channels_ = 4;
+  SetFrameData(4, 2, 6, 8, &frame_);
+
+  target_frame.num_channels_ = 2;
+  target_frame.samples_per_channel_ = frame_.samples_per_channel_;
+
+  AudioFrameOperations::QuadToStereo(frame_.data(), frame_.samples_per_channel_,
+                                     target_frame.mutable_data());
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = 320;
+  stereo_frame.num_channels_ = 2;
+  SetFrameData(3, 7, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, target_frame);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToStereoDoesNotWrapAround) {
+  frame_.num_channels_ = 4;
+  SetFrameData(-32768, -32768, -32768, -32768, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::QuadToStereo(&frame_));
+
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = 320;
+  stereo_frame.num_channels_ = 2;
+  SetFrameData(-32768, -32768, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, SwapStereoChannelsSucceedsOnStereo) {
+  SetFrameData(0, 1, &frame_);
+
+  AudioFrame swapped_frame;
+  swapped_frame.samples_per_channel_ = 320;
+  swapped_frame.num_channels_ = 2;
+  SetFrameData(1, 0, &swapped_frame);
+
+  AudioFrameOperations::SwapStereoChannels(&frame_);
+  VerifyFramesAreEqual(swapped_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, SwapStereoChannelsMuted) {
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::SwapStereoChannels(&frame_);
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, SwapStereoChannelsFailsOnMono) {
+  frame_.num_channels_ = 1;
+  // Set data to "stereo", despite it being a mono frame.
+  SetFrameData(0, 1, &frame_);
+
+  AudioFrame orig_frame;
+  orig_frame.CopyFrom(frame_);
+  AudioFrameOperations::SwapStereoChannels(&frame_);
+  // Verify that no swap occurred.
+  VerifyFramesAreEqual(orig_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, MuteDisabled) {
+  SetFrameData(1000, -1000, &frame_);
+  AudioFrameOperations::Mute(&frame_, false, false);
+
+  AudioFrame muted_frame;
+  muted_frame.samples_per_channel_ = 320;
+  muted_frame.num_channels_ = 2;
+  SetFrameData(1000, -1000, &muted_frame);
+  VerifyFramesAreEqual(muted_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, MuteEnabled) {
+  SetFrameData(1000, -1000, &frame_);
+  AudioFrameOperations::Mute(&frame_, true, true);
+
+  AudioFrame muted_frame;
+  muted_frame.samples_per_channel_ = frame_.samples_per_channel_;
+  muted_frame.num_channels_ = frame_.num_channels_;
+  ASSERT_TRUE(muted_frame.muted());
+  VerifyFramesAreEqual(muted_frame, frame_);
+}
+
+// Verify that *beginning* to mute works for short and long (>128) frames, mono
+// and stereo. Beginning mute should yield a ramp down to zero.
+TEST_F(AudioFrameOperationsTest, MuteBeginMonoLong) {
+  InitFrame(&frame_, 1, 228, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, false, true);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 99));
+  EXPECT_EQ(992, GetChannelData(frame_, 0, 100));
+  EXPECT_EQ(7, GetChannelData(frame_, 0, 226));
+  EXPECT_EQ(0, GetChannelData(frame_, 0, 227));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteBeginMonoShort) {
+  InitFrame(&frame_, 1, 93, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, false, true);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  EXPECT_EQ(989, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(978, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(10, GetChannelData(frame_, 0, 91));
+  EXPECT_EQ(0, GetChannelData(frame_, 0, 92));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteBeginStereoLong) {
+  InitFrame(&frame_, 2, 228, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, false, true);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  VerifyFrameDataBounds(frame_, 1, 0, -1000);
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 99));
+  EXPECT_EQ(-1000, GetChannelData(frame_, 1, 99));
+  EXPECT_EQ(992, GetChannelData(frame_, 0, 100));
+  EXPECT_EQ(-992, GetChannelData(frame_, 1, 100));
+  EXPECT_EQ(7, GetChannelData(frame_, 0, 226));
+  EXPECT_EQ(-7, GetChannelData(frame_, 1, 226));
+  EXPECT_EQ(0, GetChannelData(frame_, 0, 227));
+  EXPECT_EQ(0, GetChannelData(frame_, 1, 227));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteBeginStereoShort) {
+  InitFrame(&frame_, 2, 93, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, false, true);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  VerifyFrameDataBounds(frame_, 1, 0, -1000);
+  EXPECT_EQ(989, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(-989, GetChannelData(frame_, 1, 0));
+  EXPECT_EQ(978, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(-978, GetChannelData(frame_, 1, 1));
+  EXPECT_EQ(10, GetChannelData(frame_, 0, 91));
+  EXPECT_EQ(-10, GetChannelData(frame_, 1, 91));
+  EXPECT_EQ(0, GetChannelData(frame_, 0, 92));
+  EXPECT_EQ(0, GetChannelData(frame_, 1, 92));
+}
+
+// Verify that *ending* to mute works for short and long (>128) frames, mono
+// and stereo. Ending mute should yield a ramp up from zero.
+TEST_F(AudioFrameOperationsTest, MuteEndMonoLong) {
+  InitFrame(&frame_, 1, 228, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, true, false);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  EXPECT_EQ(7, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(15, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 127));
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 128));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteEndMonoShort) {
+  InitFrame(&frame_, 1, 93, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, true, false);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  EXPECT_EQ(10, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(21, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(989, GetChannelData(frame_, 0, 91));
+  EXPECT_EQ(999, GetChannelData(frame_, 0, 92));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteEndStereoLong) {
+  InitFrame(&frame_, 2, 228, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, true, false);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  VerifyFrameDataBounds(frame_, 1, 0, -1000);
+  EXPECT_EQ(7, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(-7, GetChannelData(frame_, 1, 0));
+  EXPECT_EQ(15, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(-15, GetChannelData(frame_, 1, 1));
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 127));
+  EXPECT_EQ(-1000, GetChannelData(frame_, 1, 127));
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 128));
+  EXPECT_EQ(-1000, GetChannelData(frame_, 1, 128));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteEndStereoShort) {
+  InitFrame(&frame_, 2, 93, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, true, false);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  VerifyFrameDataBounds(frame_, 1, 0, -1000);
+  EXPECT_EQ(10, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(-10, GetChannelData(frame_, 1, 0));
+  EXPECT_EQ(21, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(-21, GetChannelData(frame_, 1, 1));
+  EXPECT_EQ(989, GetChannelData(frame_, 0, 91));
+  EXPECT_EQ(-989, GetChannelData(frame_, 1, 91));
+  EXPECT_EQ(999, GetChannelData(frame_, 0, 92));
+  EXPECT_EQ(-999, GetChannelData(frame_, 1, 92));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteBeginAlreadyMuted) {
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::Mute(&frame_, false, true);
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, MuteEndAlreadyMuted) {
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::Mute(&frame_, true, false);
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, ApplyHalfGainSucceeds) {
+  SetFrameData(2, &frame_);
+
+  AudioFrame half_gain_frame;
+  half_gain_frame.num_channels_ = frame_.num_channels_;
+  half_gain_frame.samples_per_channel_ = frame_.samples_per_channel_;
+  SetFrameData(1, &half_gain_frame);
+
+  AudioFrameOperations::ApplyHalfGain(&frame_);
+  VerifyFramesAreEqual(half_gain_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, ApplyHalfGainMuted) {
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::ApplyHalfGain(&frame_);
+  EXPECT_TRUE(frame_.muted());
+}
+
+// TODO(andrew): should not allow negative scales.
+TEST_F(AudioFrameOperationsTest, DISABLED_ScaleFailsWithBadParameters) {
+  frame_.num_channels_ = 1;
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, 1.0, &frame_));
+
+  frame_.num_channels_ = 3;
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, 1.0, &frame_));
+
+  frame_.num_channels_ = 2;
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(-1.0, 1.0, &frame_));
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, -1.0, &frame_));
+}
+
+// TODO(andrew): fix the wraparound bug. We should always saturate.
+TEST_F(AudioFrameOperationsTest, DISABLED_ScaleDoesNotWrapAround) {
+  SetFrameData(4000, -4000, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::Scale(10.0, 10.0, &frame_));
+
+  AudioFrame clipped_frame;
+  clipped_frame.samples_per_channel_ = 320;
+  clipped_frame.num_channels_ = 2;
+  SetFrameData(32767, -32768, &clipped_frame);
+  VerifyFramesAreEqual(clipped_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, ScaleSucceeds) {
+  SetFrameData(1, -1, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::Scale(2.0, 3.0, &frame_));
+
+  AudioFrame scaled_frame;
+  scaled_frame.samples_per_channel_ = 320;
+  scaled_frame.num_channels_ = 2;
+  SetFrameData(2, -3, &scaled_frame);
+  VerifyFramesAreEqual(scaled_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, ScaleMuted) {
+  ASSERT_TRUE(frame_.muted());
+  EXPECT_EQ(0, AudioFrameOperations::Scale(2.0, 3.0, &frame_));
+  EXPECT_TRUE(frame_.muted());
+}
+
+// TODO(andrew): should fail with a negative scale.
+TEST_F(AudioFrameOperationsTest, DISABLED_ScaleWithSatFailsWithBadParameters) {
+  EXPECT_EQ(-1, AudioFrameOperations::ScaleWithSat(-1.0, &frame_));
+}
+
+TEST_F(AudioFrameOperationsTest, ScaleWithSatDoesNotWrapAround) {
+  frame_.num_channels_ = 1;
+  SetFrameData(4000, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(10.0, &frame_));
+
+  AudioFrame clipped_frame;
+  clipped_frame.samples_per_channel_ = 320;
+  clipped_frame.num_channels_ = 1;
+  SetFrameData(32767, &clipped_frame);
+  VerifyFramesAreEqual(clipped_frame, frame_);
+
+  SetFrameData(-4000, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(10.0, &frame_));
+  SetFrameData(-32768, &clipped_frame);
+  VerifyFramesAreEqual(clipped_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, ScaleWithSatSucceeds) {
+  frame_.num_channels_ = 1;
+  SetFrameData(1, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(2.0, &frame_));
+
+  AudioFrame scaled_frame;
+  scaled_frame.samples_per_channel_ = 320;
+  scaled_frame.num_channels_ = 1;
+  SetFrameData(2, &scaled_frame);
+  VerifyFramesAreEqual(scaled_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, ScaleWithSatMuted) {
+  ASSERT_TRUE(frame_.muted());
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(2.0, &frame_));
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, AddingXToEmptyGivesX) {
+  // When samples_per_channel_ is 0, the frame counts as empty and zero.
+  AudioFrame frame_to_add_to;
+  frame_to_add_to.mutable_data();  // Unmute the frame.
+  ASSERT_FALSE(frame_to_add_to.muted());
+  frame_to_add_to.samples_per_channel_ = 0;
+  frame_to_add_to.num_channels_ = frame_.num_channels_;
+
+  SetFrameData(1000, &frame_);
+  AudioFrameOperations::Add(frame_, &frame_to_add_to);
+  VerifyFramesAreEqual(frame_, frame_to_add_to);
+}
+
+TEST_F(AudioFrameOperationsTest, AddingXToMutedGivesX) {
+  AudioFrame frame_to_add_to;
+  ASSERT_TRUE(frame_to_add_to.muted());
+  frame_to_add_to.samples_per_channel_ = frame_.samples_per_channel_;
+  frame_to_add_to.num_channels_ = frame_.num_channels_;
+
+  SetFrameData(1000, &frame_);
+  AudioFrameOperations::Add(frame_, &frame_to_add_to);
+  VerifyFramesAreEqual(frame_, frame_to_add_to);
+}
+
+TEST_F(AudioFrameOperationsTest, AddingMutedToXGivesX) {
+  AudioFrame frame_to_add_to;
+  frame_to_add_to.samples_per_channel_ = frame_.samples_per_channel_;
+  frame_to_add_to.num_channels_ = frame_.num_channels_;
+  SetFrameData(1000, &frame_to_add_to);
+
+  AudioFrame frame_copy;
+  frame_copy.CopyFrom(frame_to_add_to);
+
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::Add(frame_, &frame_to_add_to);
+  VerifyFramesAreEqual(frame_copy, frame_to_add_to);
+}
+
+TEST_F(AudioFrameOperationsTest, AddingTwoFramesProducesTheirSum) {
+  AudioFrame frame_to_add_to;
+  frame_to_add_to.samples_per_channel_ = frame_.samples_per_channel_;
+  frame_to_add_to.num_channels_ = frame_.num_channels_;
+  SetFrameData(1000, &frame_to_add_to);
+  SetFrameData(2000, &frame_);
+
+  AudioFrameOperations::Add(frame_, &frame_to_add_to);
+  SetFrameData(frame_.data()[0] + 1000, &frame_);
+  VerifyFramesAreEqual(frame_, frame_to_add_to);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/channel_mixer.cc b/third_party/libwebrtc/audio/utility/channel_mixer.cc
new file mode 100644
index 0000000000..0f1e663873
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixer.cc
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/channel_mixer.h"
+
+#include "audio/utility/channel_mixing_matrix.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+ChannelMixer::ChannelMixer(ChannelLayout input_layout,
+                           ChannelLayout output_layout)
+    : input_layout_(input_layout),
+      output_layout_(output_layout),
+      input_channels_(ChannelLayoutToChannelCount(input_layout)),
+      output_channels_(ChannelLayoutToChannelCount(output_layout)) {
+  // Create the transformation matrix.
+  ChannelMixingMatrix matrix_builder(input_layout_, input_channels_,
+                                     output_layout_, output_channels_);
+  remapping_ = matrix_builder.CreateTransformationMatrix(&matrix_);
+}
+
+ChannelMixer::~ChannelMixer() = default;
+
+void ChannelMixer::Transform(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  RTC_DCHECK_EQ(matrix_[0].size(), static_cast<size_t>(input_channels_));
+  RTC_DCHECK_EQ(matrix_.size(), static_cast<size_t>(output_channels_));
+
+  // Leave the audio frame intact if the channel layouts for in and out are
+  // identical.
+  if (input_layout_ == output_layout_) {
+    return;
+  }
+
+  if (IsUpMixing()) {
+    RTC_CHECK_LE(frame->samples_per_channel() * output_channels_,
+                 frame->max_16bit_samples());
+  }
+
+  // Only change the number of output channels if the audio frame is muted.
+  if (frame->muted()) {
+    frame->num_channels_ = output_channels_;
+    frame->channel_layout_ = output_layout_;
+    return;
+  }
+
+  const int16_t* in_audio = frame->data();
+
+  // Only allocate fresh memory at first access or if the required size has
+  // increased.
+  // TODO(henrika): we might be able to do downmixing in-place and thereby avoid
+  // extra memory allocation and a memcpy.
+  const size_t num_elements = frame->samples_per_channel() * output_channels_;
+  if (audio_vector_ == nullptr || num_elements > audio_vector_size_) {
+    audio_vector_.reset(new int16_t[num_elements]);
+    audio_vector_size_ = num_elements;
+  }
+  int16_t* out_audio = audio_vector_.get();
+
+  // Modify the number of channels by creating a weighted sum of input samples
+  // where the weights (scale factors) for each output sample are given by the
+  // transformation matrix.
+  for (size_t i = 0; i < frame->samples_per_channel(); i++) {
+    for (size_t output_ch = 0; output_ch < output_channels_; ++output_ch) {
+      float acc_value = 0.0f;
+      for (size_t input_ch = 0; input_ch < input_channels_; ++input_ch) {
+        const float scale = matrix_[output_ch][input_ch];
+        // Scale should always be positive.
+        RTC_DCHECK_GE(scale, 0);
+        // Each output sample is a weighted sum of input samples.
+        acc_value += scale * in_audio[i * input_channels_ + input_ch];
+      }
+      const size_t index = output_channels_ * i + output_ch;
+      RTC_CHECK_LE(index, audio_vector_size_);
+      out_audio[index] = rtc::saturated_cast<int16_t>(acc_value);
+    }
+  }
+
+  // Update channel information.
+  frame->num_channels_ = output_channels_;
+  frame->channel_layout_ = output_layout_;
+
+  // Copy the output result to the audio frame in `frame`.
+  memcpy(
+      frame->mutable_data(), out_audio,
+      sizeof(int16_t) * frame->samples_per_channel() * frame->num_channels());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/channel_mixer.h b/third_party/libwebrtc/audio/utility/channel_mixer.h
new file mode 100644
index 0000000000..2dea8eb45b
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixer.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_UTILITY_CHANNEL_MIXER_H_
+#define AUDIO_UTILITY_CHANNEL_MIXER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio/channel_layout.h"
+
+namespace webrtc {
+
+// ChannelMixer is for converting audio between channel layouts.  The conversion
+// matrix is built upon construction and used during each Transform() call.  The
+// algorithm works by generating a conversion matrix mapping each output channel
+// to list of input channels.  The transform renders all of the output channels,
+// with each output channel rendered according to a weighted sum of the relevant
+// input channels as defined in the matrix.
+// This file is derived from Chromium's media/base/channel_mixer.h.
+class ChannelMixer {
+ public:
+  // To mix two channels into one and preserve loudness, we must apply
+  // (1 / sqrt(2)) gain to each.
+  static constexpr float kHalfPower = 0.707106781186547524401f;
+
+  ChannelMixer(ChannelLayout input_layout, ChannelLayout output_layout);
+  ~ChannelMixer();
+
+  // Transforms all input channels corresponding to the selected `input_layout`
+  // to the number of channels in the selected `output_layout`.
+  // Example usage (downmix from stereo to mono):
+  //
+  //   ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_MONO);
+  //   AudioFrame frame;
+  //   frame.samples_per_channel_ = 160;
+  //   frame.num_channels_ = 2;
+  //   EXPECT_EQ(2u, frame.channels());
+  //   mixer.Transform(&frame);
+  //   EXPECT_EQ(1u, frame.channels());
+  //
+  void Transform(AudioFrame* frame);
+
+ private:
+  bool IsUpMixing() const { return output_channels_ > input_channels_; }
+
+  // Selected channel layouts.
+  const ChannelLayout input_layout_;
+  const ChannelLayout output_layout_;
+
+  // Channel counts for input and output.
+  const size_t input_channels_;
+  const size_t output_channels_;
+
+  // 2D matrix of output channels to input channels.
+  std::vector<std::vector<float> > matrix_;
+
+  // 1D array used as temporary storage during the transformation.
+  std::unique_ptr<int16_t[]> audio_vector_;
+
+  // Number of elements allocated for `audio_vector_`.
+  size_t audio_vector_size_ = 0;
+
+  // Optimization case for when we can simply remap the input channels to output
+  // channels, i.e., when all scaling factors in `matrix_` equals 1.0.
+  bool remapping_;
+
+  // Delete the copy constructor and assignment operator.
+  ChannelMixer(const ChannelMixer& other) = delete;
+  ChannelMixer& operator=(const ChannelMixer& other) = delete;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_UTILITY_CHANNEL_MIXER_H_
diff --git a/third_party/libwebrtc/audio/utility/channel_mixer_unittest.cc b/third_party/libwebrtc/audio/utility/channel_mixer_unittest.cc
new file mode 100644
index 0000000000..94cb1ac7e3
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixer_unittest.cc
@@ -0,0 +1,393 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/channel_mixer.h"
+
+#include <memory>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio/channel_layout.h"
+#include "audio/utility/channel_mixing_matrix.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr uint32_t kTimestamp = 27;
+constexpr int kSampleRateHz = 16000;
+constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
+
+class ChannelMixerTest : public ::testing::Test {
+ protected:
+  ChannelMixerTest() {
+    // Use 10ms audio frames by default. Don't set values yet.
+    frame_.samples_per_channel_ = kSamplesPerChannel;
+    frame_.sample_rate_hz_ = kSampleRateHz;
+    EXPECT_TRUE(frame_.muted());
+  }
+
+  virtual ~ChannelMixerTest() {}
+
+  AudioFrame frame_;
+};
+
+void SetFrameData(int16_t data, AudioFrame* frame) {
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel() * frame->num_channels();
+       i++) {
+    frame_data[i] = data;
+  }
+}
+
+void SetMonoData(int16_t center, AudioFrame* frame) {
+  frame->num_channels_ = 1;
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel(); ++i) {
+    frame_data[i] = center;
+  }
+  EXPECT_FALSE(frame->muted());
+}
+
+void SetStereoData(int16_t left, int16_t right, AudioFrame* frame) {
+  ASSERT_LE(2 * frame->samples_per_channel(), frame->max_16bit_samples());
+  frame->num_channels_ = 2;
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel() * 2; i += 2) {
+    frame_data[i] = left;
+    frame_data[i + 1] = right;
+  }
+  EXPECT_FALSE(frame->muted());
+}
+
+void SetFiveOneData(int16_t front_left,
+                    int16_t front_right,
+                    int16_t center,
+                    int16_t lfe,
+                    int16_t side_left,
+                    int16_t side_right,
+                    AudioFrame* frame) {
+  ASSERT_LE(6 * frame->samples_per_channel(), frame->max_16bit_samples());
+  frame->num_channels_ = 6;
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel() * 6; i += 6) {
+    frame_data[i] = front_left;
+    frame_data[i + 1] = front_right;
+    frame_data[i + 2] = center;
+    frame_data[i + 3] = lfe;
+    frame_data[i + 4] = side_left;
+    frame_data[i + 5] = side_right;
+  }
+  EXPECT_FALSE(frame->muted());
+}
+
+void SetSevenOneData(int16_t front_left,
+                     int16_t front_right,
+                     int16_t center,
+                     int16_t lfe,
+                     int16_t side_left,
+                     int16_t side_right,
+                     int16_t back_left,
+                     int16_t back_right,
+                     AudioFrame* frame) {
+  ASSERT_LE(8 * frame->samples_per_channel(), frame->max_16bit_samples());
+  frame->num_channels_ = 8;
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel() * 8; i += 8) {
+    frame_data[i] = front_left;
+    frame_data[i + 1] = front_right;
+    frame_data[i + 2] = center;
+    frame_data[i + 3] = lfe;
+    frame_data[i + 4] = side_left;
+    frame_data[i + 5] = side_right;
+    frame_data[i + 6] = back_left;
+    frame_data[i + 7] = back_right;
+  }
+  EXPECT_FALSE(frame->muted());
+}
+
+bool AllSamplesEquals(int16_t sample, const AudioFrame* frame) {
+  const int16_t* frame_data = frame->data();
+  for (size_t i = 0; i < frame->samples_per_channel() * frame->num_channels();
+       i++) {
+    if (frame_data[i] != sample) {
+      return false;
+    }
+  }
+  return true;
+}
+
+void VerifyFramesAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
+  EXPECT_EQ(frame1.num_channels(), frame2.num_channels());
+  EXPECT_EQ(frame1.samples_per_channel(), frame2.samples_per_channel());
+  const int16_t* frame1_data = frame1.data();
+  const int16_t* frame2_data = frame2.data();
+  for (size_t i = 0; i < frame1.samples_per_channel() * frame1.num_channels();
+       i++) {
+    EXPECT_EQ(frame1_data[i], frame2_data[i]);
+  }
+  EXPECT_EQ(frame1.muted(), frame2.muted());
+}
+
+}  // namespace
+
+// Test all possible layout conversions can be constructed and mixed. Don't
+// care about the actual content, simply run through all mixing combinations
+// and ensure that nothing fails.
+TEST_F(ChannelMixerTest, ConstructAllPossibleLayouts) {
+  for (ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+       input_layout <= CHANNEL_LAYOUT_MAX;
+       input_layout = static_cast<ChannelLayout>(input_layout + 1)) {
+    for (ChannelLayout output_layout = CHANNEL_LAYOUT_MONO;
+         output_layout <= CHANNEL_LAYOUT_MAX;
+         output_layout = static_cast<ChannelLayout>(output_layout + 1)) {
+      // DISCRETE, BITSTREAM can't be tested here based on the current approach.
+      // CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC is not mixable.
+      // Stereo down mix should never be the output layout.
+      if (input_layout == CHANNEL_LAYOUT_BITSTREAM ||
+          input_layout == CHANNEL_LAYOUT_DISCRETE ||
+          input_layout == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC ||
+          output_layout == CHANNEL_LAYOUT_BITSTREAM ||
+          output_layout == CHANNEL_LAYOUT_DISCRETE ||
+          output_layout == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC ||
+          output_layout == CHANNEL_LAYOUT_STEREO_DOWNMIX) {
+        continue;
+      }
+
+      rtc::StringBuilder ss;
+      ss << "Input Layout: " << input_layout
+         << ", Output Layout: " << output_layout;
+      SCOPED_TRACE(ss.str());
+      ChannelMixer mixer(input_layout, output_layout);
+
+      frame_.UpdateFrame(kTimestamp, nullptr, kSamplesPerChannel, kSampleRateHz,
+                         AudioFrame::kNormalSpeech, AudioFrame::kVadActive,
+                         ChannelLayoutToChannelCount(input_layout));
+      EXPECT_TRUE(frame_.muted());
+      mixer.Transform(&frame_);
+    }
+  }
+}
+
+// Ensure that the audio frame is untouched when input and output channel
+// layouts are identical, i.e., the transformation should have no effect.
+// Exclude invalid mixing combinations.
+TEST_F(ChannelMixerTest, NoMixingForIdenticalChannelLayouts) {
+  for (ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+       input_layout <= CHANNEL_LAYOUT_MAX;
+       input_layout = static_cast<ChannelLayout>(input_layout + 1)) {
+    for (ChannelLayout output_layout = CHANNEL_LAYOUT_MONO;
+         output_layout <= CHANNEL_LAYOUT_MAX;
+         output_layout = static_cast<ChannelLayout>(output_layout + 1)) {
+      if (input_layout != output_layout ||
+          input_layout == CHANNEL_LAYOUT_BITSTREAM ||
+          input_layout == CHANNEL_LAYOUT_DISCRETE ||
+          input_layout == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC ||
+          output_layout == CHANNEL_LAYOUT_STEREO_DOWNMIX) {
+        continue;
+      }
+      ChannelMixer mixer(input_layout, output_layout);
+      frame_.num_channels_ = ChannelLayoutToChannelCount(input_layout);
+      SetFrameData(99, &frame_);
+      mixer.Transform(&frame_);
+      EXPECT_EQ(ChannelLayoutToChannelCount(input_layout),
+                static_cast<int>(frame_.num_channels()));
+      EXPECT_TRUE(AllSamplesEquals(99, &frame_));
+    }
+  }
+}
+
+TEST_F(ChannelMixerTest, StereoToMono) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_MONO);
+  //
+  //                      Input: stereo
+  //                      LEFT  RIGHT
+  // Output: mono CENTER  0.5   0.5
+  //
+  SetStereoData(7, 3, &frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame_.channel_layout());
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetMonoData(5, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+
+  SetStereoData(-32768, -32768, &frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame_.channel_layout());
+  SetMonoData(-32768, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(ChannelMixerTest, StereoToMonoMuted) {
+  ASSERT_TRUE(frame_.muted());
+  ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_MONO);
+  mixer.Transform(&frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame_.channel_layout());
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(ChannelMixerTest, FiveOneToSevenOneMuted) {
+  ASSERT_TRUE(frame_.muted());
+  ChannelMixer mixer(CHANNEL_LAYOUT_5_1, CHANNEL_LAYOUT_7_1);
+  mixer.Transform(&frame_);
+  EXPECT_EQ(8u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_7_1, frame_.channel_layout());
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(ChannelMixerTest, FiveOneToMono) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_5_1, CHANNEL_LAYOUT_MONO);
+  //
+  //                      Input: 5.1
+  //                      LEFT   RIGHT  CENTER  LFE    SIDE_LEFT  SIDE_RIGHT
+  // Output: mono CENTER  0.707  0.707  1       0.707  0.707      0.707
+  //
+  // a = [10, 20, 15, 2, 5, 5]
+  // b = [1/sqrt(2), 1/sqrt(2), 1.0, 1/sqrt(2), 1/sqrt(2), 1/sqrt(2)] =>
+  // a * b (dot product) = 44.69848480983499,
+  // which is truncated into 44 using 16 bit representation.
+  //
+  SetFiveOneData(10, 20, 15, 2, 5, 5, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame_.channel_layout());
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetMonoData(44, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+
+  SetFiveOneData(-32768, -32768, -32768, -32768, -32768, -32768, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame_.channel_layout());
+  SetMonoData(-32768, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(ChannelMixerTest, FiveOneToSevenOne) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_5_1, CHANNEL_LAYOUT_7_1);
+  //
+  //                        Input: 5.1
+  //                        LEFT   RIGHT  CENTER  LFE    SIDE_LEFT  SIDE_RIGHT
+  // Output: 7.1 LEFT       1      0      0       0      0          0
+  //             RIGHT      0      1      0       0      0          0
+  //             CENTER     0      0      1       0      0          0
+  //             LFE        0      0      0       1      0          0
+  //             SIDE_LEFT  0      0      0       0      1          0
+  //             SIDE_RIGHT 0      0      0       0      0          1
+  //             BACK_LEFT  0      0      0       0      0          0
+  //             BACK_RIGHT 0      0      0       0      0          0
+  //
+  SetFiveOneData(10, 20, 15, 2, 5, 5, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(8u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_7_1, frame_.channel_layout());
+
+  AudioFrame seven_one_frame;
+  seven_one_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetSevenOneData(10, 20, 15, 2, 5, 5, 0, 0, &seven_one_frame);
+  VerifyFramesAreEqual(seven_one_frame, frame_);
+
+  SetFiveOneData(-32768, 32767, -32768, 32767, -32768, 32767, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(8u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_7_1, frame_.channel_layout());
+  SetSevenOneData(-32768, 32767, -32768, 32767, -32768, 32767, 0, 0,
+                  &seven_one_frame);
+  VerifyFramesAreEqual(seven_one_frame, frame_);
+}
+
+TEST_F(ChannelMixerTest, FiveOneBackToStereo) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_5_1_BACK, CHANNEL_LAYOUT_STEREO);
+  //
+  //                      Input: 5.1
+  //                      LEFT   RIGHT  CENTER  LFE    BACK_LEFT  BACK_RIGHT
+  // Output: stereo LEFT  1      0      0.707   0.707  0.707      0
+  //                RIGHT 0      1      0.707   0.707  0          0.707
+  //
+  SetFiveOneData(20, 30, 15, 2, 5, 5, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame_.channel_layout());
+
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetStereoData(35, 45, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+
+  SetFiveOneData(-32768, -32768, -32768, -32768, -32768, -32768, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame_.channel_layout());
+  SetStereoData(-32768, -32768, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+}
+
+TEST_F(ChannelMixerTest, MonoToStereo) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_MONO, CHANNEL_LAYOUT_STEREO);
+  //
+  //                       Input: mono
+  //                       CENTER
+  // Output: stereo LEFT   1
+  //                RIGHT  1
+  //
+  SetMonoData(44, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame_.channel_layout());
+
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetStereoData(44, 44, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+}
+
+TEST_F(ChannelMixerTest, StereoToFiveOne) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_5_1);
+  //
+  //                         Input: Stereo
+  //                         LEFT   RIGHT
+  // Output: 5.1 LEFT        1      0
+  //             RIGHT       0      1
+  //             CENTER      0      0
+  //             LFE         0      0
+  //             SIDE_LEFT   0      0
+  //             SIDE_RIGHT  0      0
+  //
+  SetStereoData(50, 60, &frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_5_1, frame_.channel_layout());
+
+  AudioFrame five_one_frame;
+  five_one_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetFiveOneData(50, 60, 0, 0, 0, 0, &five_one_frame);
+  VerifyFramesAreEqual(five_one_frame, frame_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/channel_mixing_matrix.cc b/third_party/libwebrtc/audio/utility/channel_mixing_matrix.cc
new file mode 100644
index 0000000000..1244653f63
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixing_matrix.cc
@@ -0,0 +1,333 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/channel_mixing_matrix.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+
+#include "audio/utility/channel_mixer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+// Selects the default usage of VoIP channel mapping adjustments.
+bool UseChannelMappingAdjustmentsByDefault() {
+  return !field_trial::IsEnabled(
+      "WebRTC-VoIPChannelRemixingAdjustmentKillSwitch");
+}
+
+}  // namespace
+
+static void ValidateLayout(ChannelLayout layout) {
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_NONE);
+  RTC_CHECK_LE(layout, CHANNEL_LAYOUT_MAX);
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_UNSUPPORTED);
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_DISCRETE);
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC);
+
+  // Verify there's at least one channel.  Should always be true here by virtue
+  // of not being one of the invalid layouts, but lets double check to be sure.
+  int channel_count = ChannelLayoutToChannelCount(layout);
+  RTC_DCHECK_GT(channel_count, 0);
+
+  // If we have more than one channel, verify a symmetric layout for sanity.
+  // The unit test will verify all possible layouts, so this can be a DCHECK.
+  // Symmetry allows simplifying the matrix building code by allowing us to
+  // assume that if one channel of a pair exists, the other will too.
+  if (channel_count > 1) {
+    // Assert that LEFT exists if and only if RIGHT exists, and so on.
+    RTC_DCHECK_EQ(ChannelOrder(layout, LEFT) >= 0,
+                  ChannelOrder(layout, RIGHT) >= 0);
+    RTC_DCHECK_EQ(ChannelOrder(layout, SIDE_LEFT) >= 0,
+                  ChannelOrder(layout, SIDE_RIGHT) >= 0);
+    RTC_DCHECK_EQ(ChannelOrder(layout, BACK_LEFT) >= 0,
+                  ChannelOrder(layout, BACK_RIGHT) >= 0);
+    RTC_DCHECK_EQ(ChannelOrder(layout, LEFT_OF_CENTER) >= 0,
+                  ChannelOrder(layout, RIGHT_OF_CENTER) >= 0);
+  } else {
+    RTC_DCHECK_EQ(layout, CHANNEL_LAYOUT_MONO);
+  }
+}
+
+ChannelMixingMatrix::ChannelMixingMatrix(ChannelLayout input_layout,
+                                         int input_channels,
+                                         ChannelLayout output_layout,
+                                         int output_channels)
+    : use_voip_channel_mapping_adjustments_(
+          UseChannelMappingAdjustmentsByDefault()),
+      input_layout_(input_layout),
+      input_channels_(input_channels),
+      output_layout_(output_layout),
+      output_channels_(output_channels) {
+  // Stereo down mix should never be the output layout.
+  RTC_CHECK_NE(output_layout, CHANNEL_LAYOUT_STEREO_DOWNMIX);
+
+  // Verify that the layouts are supported
+  if (input_layout != CHANNEL_LAYOUT_DISCRETE)
+    ValidateLayout(input_layout);
+  if (output_layout != CHANNEL_LAYOUT_DISCRETE)
+    ValidateLayout(output_layout);
+
+  // Special case for 5.0, 5.1 with back channels when upmixed to 7.0, 7.1,
+  // which should map the back LR to side LR.
+  if (input_layout_ == CHANNEL_LAYOUT_5_0_BACK &&
+      output_layout_ == CHANNEL_LAYOUT_7_0) {
+    input_layout_ = CHANNEL_LAYOUT_5_0;
+  } else if (input_layout_ == CHANNEL_LAYOUT_5_1_BACK &&
+             output_layout_ == CHANNEL_LAYOUT_7_1) {
+    input_layout_ = CHANNEL_LAYOUT_5_1;
+  }
+}
+
+ChannelMixingMatrix::~ChannelMixingMatrix() = default;
+
+bool ChannelMixingMatrix::CreateTransformationMatrix(
+    std::vector<std::vector<float>>* matrix) {
+  matrix_ = matrix;
+
+  // Size out the initial matrix.
+  matrix_->reserve(output_channels_);
+  for (int output_ch = 0; output_ch < output_channels_; ++output_ch)
+    matrix_->push_back(std::vector<float>(input_channels_, 0));
+
+  // First check for discrete case.
+  if (input_layout_ == CHANNEL_LAYOUT_DISCRETE ||
+      output_layout_ == CHANNEL_LAYOUT_DISCRETE) {
+    // If the number of input channels is more than output channels, then
+    // copy as many as we can then drop the remaining input channels.
+    // If the number of input channels is less than output channels, then
+    // copy them all, then zero out the remaining output channels.
+    int passthrough_channels = std::min(input_channels_, output_channels_);
+    for (int i = 0; i < passthrough_channels; ++i)
+      (*matrix_)[i][i] = 1;
+
+    return true;
+  }
+
+  // If specified, use adjusted channel mapping for the VoIP scenario.
+  if (use_voip_channel_mapping_adjustments_ &&
+      input_layout_ == CHANNEL_LAYOUT_MONO &&
+      ChannelLayoutToChannelCount(output_layout_) >= 2) {
+    // Only place the mono input in the front left and right channels.
+    (*matrix_)[0][0] = 1.f;
+    (*matrix_)[1][0] = 1.f;
+
+    for (size_t output_ch = 2; output_ch < matrix_->size(); ++output_ch) {
+      (*matrix_)[output_ch][0] = 0.f;
+    }
+    return true;
+  }
+
+  // Route matching channels and figure out which ones aren't accounted for.
+  for (Channels ch = LEFT; ch < CHANNELS_MAX + 1;
+       ch = static_cast<Channels>(ch + 1)) {
+    int input_ch_index = ChannelOrder(input_layout_, ch);
+    if (input_ch_index < 0)
+      continue;
+
+    int output_ch_index = ChannelOrder(output_layout_, ch);
+    if (output_ch_index < 0) {
+      unaccounted_inputs_.push_back(ch);
+      continue;
+    }
+
+    RTC_DCHECK_LT(static_cast<size_t>(output_ch_index), matrix_->size());
+    RTC_DCHECK_LT(static_cast<size_t>(input_ch_index),
+                  (*matrix_)[output_ch_index].size());
+    (*matrix_)[output_ch_index][input_ch_index] = 1;
+  }
+
+  // If all input channels are accounted for, there's nothing left to do.
+  if (unaccounted_inputs_.empty()) {
+    // Since all output channels map directly to inputs we can optimize.
+    return true;
+  }
+
+  // Mix front LR into center.
+  if (IsUnaccounted(LEFT)) {
+    // When down mixing to mono from stereo, we need to be careful of full scale
+    // stereo mixes.  Scaling by 1 / sqrt(2) here will likely lead to clipping
+    // so we use 1 / 2 instead.
+    float scale =
+        (output_layout_ == CHANNEL_LAYOUT_MONO && input_channels_ == 2)
+            ? 0.5
+            : ChannelMixer::kHalfPower;
+    Mix(LEFT, CENTER, scale);
+    Mix(RIGHT, CENTER, scale);
+  }
+
+  // Mix center into front LR.
+  if (IsUnaccounted(CENTER)) {
+    // When up mixing from mono, just do a copy to front LR.
+    float scale =
+        (input_layout_ == CHANNEL_LAYOUT_MONO) ? 1 : ChannelMixer::kHalfPower;
+    MixWithoutAccounting(CENTER, LEFT, scale);
+    Mix(CENTER, RIGHT, scale);
+  }
+
+  // Mix back LR into: side LR || back center || front LR || front center.
+  if (IsUnaccounted(BACK_LEFT)) {
+    if (HasOutputChannel(SIDE_LEFT)) {
+      // If the input has side LR, mix back LR into side LR, but instead if the
+      // input doesn't have side LR (but output does) copy back LR to side LR.
+      float scale = HasInputChannel(SIDE_LEFT) ? ChannelMixer::kHalfPower : 1;
+      Mix(BACK_LEFT, SIDE_LEFT, scale);
+      Mix(BACK_RIGHT, SIDE_RIGHT, scale);
+    } else if (HasOutputChannel(BACK_CENTER)) {
+      // Mix back LR into back center.
+      Mix(BACK_LEFT, BACK_CENTER, ChannelMixer::kHalfPower);
+      Mix(BACK_RIGHT, BACK_CENTER, ChannelMixer::kHalfPower);
+    } else if (output_layout_ > CHANNEL_LAYOUT_MONO) {
+      // Mix back LR into front LR.
+      Mix(BACK_LEFT, LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_RIGHT, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix back LR into front center.
+      Mix(BACK_LEFT, CENTER, ChannelMixer::kHalfPower);
+      Mix(BACK_RIGHT, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix side LR into: back LR || back center || front LR || front center.
+  if (IsUnaccounted(SIDE_LEFT)) {
+    if (HasOutputChannel(BACK_LEFT)) {
+      // If the input has back LR, mix side LR into back LR, but instead if the
+      // input doesn't have back LR (but output does) copy side LR to back LR.
+      float scale = HasInputChannel(BACK_LEFT) ? ChannelMixer::kHalfPower : 1;
+      Mix(SIDE_LEFT, BACK_LEFT, scale);
+      Mix(SIDE_RIGHT, BACK_RIGHT, scale);
+    } else if (HasOutputChannel(BACK_CENTER)) {
+      // Mix side LR into back center.
+      Mix(SIDE_LEFT, BACK_CENTER, ChannelMixer::kHalfPower);
+      Mix(SIDE_RIGHT, BACK_CENTER, ChannelMixer::kHalfPower);
+    } else if (output_layout_ > CHANNEL_LAYOUT_MONO) {
+      // Mix side LR into front LR.
+      Mix(SIDE_LEFT, LEFT, ChannelMixer::kHalfPower);
+      Mix(SIDE_RIGHT, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix side LR into front center.
+      Mix(SIDE_LEFT, CENTER, ChannelMixer::kHalfPower);
+      Mix(SIDE_RIGHT, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix back center into: back LR || side LR || front LR || front center.
+  if (IsUnaccounted(BACK_CENTER)) {
+    if (HasOutputChannel(BACK_LEFT)) {
+      // Mix back center into back LR.
+      MixWithoutAccounting(BACK_CENTER, BACK_LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_CENTER, BACK_RIGHT, ChannelMixer::kHalfPower);
+    } else if (HasOutputChannel(SIDE_LEFT)) {
+      // Mix back center into side LR.
+      MixWithoutAccounting(BACK_CENTER, SIDE_LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_CENTER, SIDE_RIGHT, ChannelMixer::kHalfPower);
+    } else if (output_layout_ > CHANNEL_LAYOUT_MONO) {
+      // Mix back center into front LR.
+      // TODO(dalecurtis): Not sure about these values?
+      MixWithoutAccounting(BACK_CENTER, LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_CENTER, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix back center into front center.
+      // TODO(dalecurtis): Not sure about these values?
+      Mix(BACK_CENTER, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix LR of center into: front LR || front center.
+  if (IsUnaccounted(LEFT_OF_CENTER)) {
+    if (HasOutputChannel(LEFT)) {
+      // Mix LR of center into front LR.
+      Mix(LEFT_OF_CENTER, LEFT, ChannelMixer::kHalfPower);
+      Mix(RIGHT_OF_CENTER, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix LR of center into front center.
+      Mix(LEFT_OF_CENTER, CENTER, ChannelMixer::kHalfPower);
+      Mix(RIGHT_OF_CENTER, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix LFE into: front center || front LR.
+  if (IsUnaccounted(LFE)) {
+    if (!HasOutputChannel(CENTER)) {
+      // Mix LFE into front LR.
+      MixWithoutAccounting(LFE, LEFT, ChannelMixer::kHalfPower);
+      Mix(LFE, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix LFE into front center.
+      Mix(LFE, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // All channels should now be accounted for.
+  RTC_DCHECK(unaccounted_inputs_.empty());
+
+  // See if the output `matrix_` is simply a remapping matrix.  If each input
+  // channel maps to a single output channel we can simply remap.  Doing this
+  // programmatically is less fragile than logic checks on channel mappings.
+  for (int output_ch = 0; output_ch < output_channels_; ++output_ch) {
+    int input_mappings = 0;
+    for (int input_ch = 0; input_ch < input_channels_; ++input_ch) {
+      // We can only remap if each row contains a single scale of 1.  I.e., each
+      // output channel is mapped from a single unscaled input channel.
+      if ((*matrix_)[output_ch][input_ch] != 1 || ++input_mappings > 1)
+        return false;
+    }
+  }
+
+  // If we've gotten here, `matrix_` is simply a remapping.
+  return true;
+}
+
+void ChannelMixingMatrix::AccountFor(Channels ch) {
+  unaccounted_inputs_.erase(
+      std::find(unaccounted_inputs_.begin(), unaccounted_inputs_.end(), ch));
+}
+
+bool ChannelMixingMatrix::IsUnaccounted(Channels ch) const {
+  return std::find(unaccounted_inputs_.begin(), unaccounted_inputs_.end(),
+                   ch) != unaccounted_inputs_.end();
+}
+
+bool ChannelMixingMatrix::HasInputChannel(Channels ch) const {
+  return ChannelOrder(input_layout_, ch) >= 0;
+}
+
+bool ChannelMixingMatrix::HasOutputChannel(Channels ch) const {
+  return ChannelOrder(output_layout_, ch) >= 0;
+}
+
+void ChannelMixingMatrix::Mix(Channels input_ch,
+                              Channels output_ch,
+                              float scale) {
+  MixWithoutAccounting(input_ch, output_ch, scale);
+  AccountFor(input_ch);
+}
+
+void ChannelMixingMatrix::MixWithoutAccounting(Channels input_ch,
+                                               Channels output_ch,
+                                               float scale) {
+  int input_ch_index = ChannelOrder(input_layout_, input_ch);
+  int output_ch_index = ChannelOrder(output_layout_, output_ch);
+
+  RTC_DCHECK(IsUnaccounted(input_ch));
+  RTC_DCHECK_GE(input_ch_index, 0);
+  RTC_DCHECK_GE(output_ch_index, 0);
+
+  RTC_DCHECK_EQ((*matrix_)[output_ch_index][input_ch_index], 0);
+  (*matrix_)[output_ch_index][input_ch_index] = scale;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/channel_mixing_matrix.h b/third_party/libwebrtc/audio/utility/channel_mixing_matrix.h
new file mode 100644
index 0000000000..ee00860846
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixing_matrix.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_UTILITY_CHANNEL_MIXING_MATRIX_H_
+#define AUDIO_UTILITY_CHANNEL_MIXING_MATRIX_H_
+
+#include <vector>
+
+#include "api/audio/channel_layout.h"
+
+namespace webrtc {
+
+class ChannelMixingMatrix {
+ public:
+  ChannelMixingMatrix(ChannelLayout input_layout,
+                      int input_channels,
+                      ChannelLayout output_layout,
+                      int output_channels);
+
+  ~ChannelMixingMatrix();
+
+  // Create the transformation matrix of input channels to output channels.
+  // Updates the empty matrix with the transformation, and returns true
+  // if the transformation is just a remapping of channels (no mixing).
+  // The size of `matrix` is `output_channels` x `input_channels`, i.e., the
+  // number of rows equals the number of output channels and the number of
+  // columns corresponds to the number of input channels.
+  // This file is derived from Chromium's media/base/channel_mixing_matrix.h.
+  bool CreateTransformationMatrix(std::vector<std::vector<float>>* matrix);
+
+ private:
+  const bool use_voip_channel_mapping_adjustments_;
+
+  // Result transformation of input channels to output channels
+  std::vector<std::vector<float>>* matrix_;
+
+  // Input and output channel layout provided during construction.
+  ChannelLayout input_layout_;
+  int input_channels_;
+  ChannelLayout output_layout_;
+  int output_channels_;
+
+  // Helper variable for tracking which inputs are currently unaccounted,
+  // should be empty after construction completes.
+  std::vector<Channels> unaccounted_inputs_;
+
+  // Helper methods for managing unaccounted input channels.
+  void AccountFor(Channels ch);
+  bool IsUnaccounted(Channels ch) const;
+
+  // Helper methods for checking if `ch` exists in either `input_layout_` or
+  // `output_layout_` respectively.
+  bool HasInputChannel(Channels ch) const;
+  bool HasOutputChannel(Channels ch) const;
+
+  // Helper methods for updating `matrix_` with the proper value for
+  // mixing `input_ch` into `output_ch`.  MixWithoutAccounting() does not
+  // remove the channel from `unaccounted_inputs_`.
+  void Mix(Channels input_ch, Channels output_ch, float scale);
+  void MixWithoutAccounting(Channels input_ch, Channels output_ch, float scale);
+
+  // Delete the copy constructor and assignment operator.
+  ChannelMixingMatrix(const ChannelMixingMatrix& other) = delete;
+  ChannelMixingMatrix& operator=(const ChannelMixingMatrix& other) = delete;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_UTILITY_CHANNEL_MIXING_MATRIX_H_
diff --git a/third_party/libwebrtc/audio/utility/channel_mixing_matrix_unittest.cc b/third_party/libwebrtc/audio/utility/channel_mixing_matrix_unittest.cc
new file mode 100644
index 0000000000..a4efb4fd38
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixing_matrix_unittest.cc
@@ -0,0 +1,476 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/channel_mixing_matrix.h"
+
+#include <stddef.h>
+
+#include "audio/utility/channel_mixer.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Test all possible layout conversions can be constructed and mixed.
+// Also ensure that the channel matrix fulfill certain conditions when remapping
+// is supported.
+TEST(ChannelMixingMatrixTest, ConstructAllPossibleLayouts) {
+  for (ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+       input_layout <= CHANNEL_LAYOUT_MAX;
+       input_layout = static_cast<ChannelLayout>(input_layout + 1)) {
+    for (ChannelLayout output_layout = CHANNEL_LAYOUT_MONO;
+         output_layout <= CHANNEL_LAYOUT_MAX;
+         output_layout = static_cast<ChannelLayout>(output_layout + 1)) {
+      // DISCRETE, BITSTREAM can't be tested here based on the current approach.
+      // CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC is not mixable.
+      // Stereo down mix should never be the output layout.
+      if (input_layout == CHANNEL_LAYOUT_BITSTREAM ||
+          input_layout == CHANNEL_LAYOUT_DISCRETE ||
+          input_layout == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC ||
+          output_layout == CHANNEL_LAYOUT_BITSTREAM ||
+          output_layout == CHANNEL_LAYOUT_DISCRETE ||
+          output_layout == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC ||
+          output_layout == CHANNEL_LAYOUT_STEREO_DOWNMIX) {
+        continue;
+      }
+
+      rtc::StringBuilder ss;
+      ss << "Input Layout: " << input_layout
+         << ", Output Layout: " << output_layout;
+      SCOPED_TRACE(ss.str());
+      ChannelMixingMatrix matrix_builder(
+          input_layout, ChannelLayoutToChannelCount(input_layout),
+          output_layout, ChannelLayoutToChannelCount(output_layout));
+      const int input_channels = ChannelLayoutToChannelCount(input_layout);
+      const int output_channels = ChannelLayoutToChannelCount(output_layout);
+      std::vector<std::vector<float>> matrix;
+      bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+      if (remapping) {
+        // Also ensure that (when remapping can take place), a maximum of one
+        // input channel is included per output. This knowledge will simplify
+        // the channel mixing algorithm since it allows us to find the only
+        // scale factor which equals 1.0 and copy that input to its
+        // corresponding output. If no such factor can be found, the
+        // corresponding output can be set to zero.
+        for (int i = 0; i < output_channels; i++) {
+          EXPECT_EQ(static_cast<size_t>(input_channels), matrix[i].size());
+          int num_input_channels_accounted_for_per_output = 0;
+          for (int j = 0; j < input_channels; j++) {
+            float scale = matrix[i][j];
+            if (scale > 0) {
+              EXPECT_EQ(scale, 1.0f);
+              num_input_channels_accounted_for_per_output++;
+            }
+          }
+          // Each output channel shall contain contribution from one or less
+          // input channels.
+          EXPECT_LE(num_input_channels_accounted_for_per_output, 1);
+        }
+      }
+    }
+  }
+}
+
+// Verify channels are mixed and scaled correctly.
+TEST(ChannelMixingMatrixTest, StereoToMono) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_STEREO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_MONO;
+  ChannelMixingMatrix matrix_builder(
+      input_layout, ChannelLayoutToChannelCount(input_layout), output_layout,
+      ChannelLayoutToChannelCount(output_layout));
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                      Input: stereo
+  //                      LEFT  RIGHT
+  // Output: mono CENTER  0.5   0.5
+  //
+  EXPECT_FALSE(remapping);
+  EXPECT_EQ(1u, matrix.size());
+  EXPECT_EQ(2u, matrix[0].size());
+  EXPECT_EQ(0.5f, matrix[0][0]);
+  EXPECT_EQ(0.5f, matrix[0][1]);
+}
+
+TEST(ChannelMixingMatrixTest, MonoToStereo) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_STEREO;
+  ChannelMixingMatrix matrix_builder(
+      input_layout, ChannelLayoutToChannelCount(input_layout), output_layout,
+      ChannelLayoutToChannelCount(output_layout));
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                       Input: mono
+  //                       CENTER
+  // Output: stereo LEFT   1
+  //                RIGHT  1
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(2u, matrix.size());
+  EXPECT_EQ(1u, matrix[0].size());
+  EXPECT_EQ(1.0f, matrix[0][0]);
+  EXPECT_EQ(1u, matrix[1].size());
+  EXPECT_EQ(1.0f, matrix[1][0]);
+}
+
+TEST(ChannelMixingMatrixTest, MonoToTwoOneWithoutVoIPAdjustments) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VoIPChannelRemixingAdjustmentKillSwitch/Enabled/");
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_2_1;
+  ChannelMixingMatrix matrix_builder(
+      input_layout, ChannelLayoutToChannelCount(input_layout), output_layout,
+      ChannelLayoutToChannelCount(output_layout));
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                         Input: mono
+  //                         CENTER
+  // Output: 2.1 FRONT_LEFT  1
+  //             FRONT_RIGHT 1
+  //             BACK_CENTER 0
+  //
+  EXPECT_FALSE(remapping);
+  EXPECT_EQ(3u, matrix.size());
+  EXPECT_EQ(1u, matrix[0].size());
+  EXPECT_EQ(1.0f, matrix[0][0]);
+  EXPECT_EQ(1.0f, matrix[1][0]);
+  EXPECT_EQ(0.0f, matrix[2][0]);
+}
+
+TEST(ChannelMixingMatrixTest, MonoToTwoOneWithVoIPAdjustments) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_2_1;
+  ChannelMixingMatrix matrix_builder(
+      input_layout, ChannelLayoutToChannelCount(input_layout), output_layout,
+      ChannelLayoutToChannelCount(output_layout));
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                         Input: mono
+  //                         CENTER
+  // Output: 2.1 FRONT_LEFT  1
+  //             FRONT_RIGHT 1
+  //             BACK_CENTER 0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(3u, matrix.size());
+  EXPECT_EQ(1u, matrix[0].size());
+  EXPECT_EQ(1.0f, matrix[0][0]);
+  EXPECT_EQ(1.0f, matrix[1][0]);
+  EXPECT_EQ(0.0f, matrix[2][0]);
+}
+
+TEST(ChannelMixingMatrixTest, MonoToFiveOneWithoutVoIPAdjustments) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VoIPChannelRemixingAdjustmentKillSwitch/Enabled/");
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_5_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+  //                         Input: mono
+  //                         CENTER
+  // Output: 5.1 LEFT        0
+  //             RIGHT       0
+  //             CENTER      1
+  //             LFE         0
+  //             SIDE_LEFT   0
+  //             SIDE_RIGHT  0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int n = 0; n < output_channels; n++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[n].size());
+    if (n == CENTER) {
+      EXPECT_EQ(1.0f, matrix[CENTER][0]);
+    } else {
+      EXPECT_EQ(0.0f, matrix[n][0]);
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, MonoToFiveOneWithVoIPAdjustments) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_5_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+  //                         Input: mono
+  //                         CENTER
+  // Output: 5.1 LEFT        1
+  //             RIGHT       1
+  //             CENTER      0
+  //             LFE         0
+  //             SIDE_LEFT   0
+  //             SIDE_RIGHT  0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int n = 0; n < output_channels; n++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[n].size());
+    if (n == LEFT || n == RIGHT) {
+      EXPECT_EQ(1.0f, matrix[n][0]);
+    } else {
+      EXPECT_EQ(0.0f, matrix[n][0]);
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, MonoToSevenOneWithoutVoIPAdjustments) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VoIPChannelRemixingAdjustmentKillSwitch/Enabled/");
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_7_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+  //                         Input: mono
+  //                         CENTER
+  // Output: 7.1 LEFT        0
+  //             RIGHT       0
+  //             CENTER      1
+  //             LFE         0
+  //             SIDE_LEFT   0
+  //             SIDE_RIGHT  0
+  //             BACK_LEFT   0
+  //             BACK_RIGHT  0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int n = 0; n < output_channels; n++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[n].size());
+    if (n == CENTER) {
+      EXPECT_EQ(1.0f, matrix[CENTER][0]);
+    } else {
+      EXPECT_EQ(0.0f, matrix[n][0]);
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, MonoToSevenOneWithVoIPAdjustments) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_7_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+  //                         Input: mono
+  //                         CENTER
+  // Output: 7.1 LEFT        1
+  //             RIGHT       1
+  //             CENTER      0
+  //             LFE         0
+  //             SIDE_LEFT   0
+  //             SIDE_RIGHT  0
+  //             BACK_LEFT   0
+  //             BACK_RIGHT  0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int n = 0; n < output_channels; n++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[n].size());
+    if (n == LEFT || n == RIGHT) {
+      EXPECT_EQ(1.0f, matrix[n][0]);
+    } else {
+      EXPECT_EQ(0.0f, matrix[n][0]);
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, FiveOneToMono) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_5_1;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_MONO;
+  ChannelMixingMatrix matrix_builder(
+      input_layout, ChannelLayoutToChannelCount(input_layout), output_layout,
+      ChannelLayoutToChannelCount(output_layout));
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  // Note: 1/sqrt(2) is shown as 0.707.
+  //
+  //                      Input: 5.1
+  //                      LEFT   RIGHT  CENTER  LFE    SIDE_LEFT  SIDE_RIGHT
+  // Output: mono CENTER  0.707  0.707  1       0.707  0.707      0.707
+  //
+  EXPECT_FALSE(remapping);
+  EXPECT_EQ(1u, matrix.size());
+  EXPECT_EQ(6u, matrix[0].size());
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[0][0]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[0][1]);
+  // The center channel will be mixed at scale 1.
+  EXPECT_EQ(1.0f, matrix[0][2]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[0][3]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[0][4]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[0][5]);
+}
+
+TEST(ChannelMixingMatrixTest, FiveOneBackToStereo) {
+  // Front L, Front R, Front C, LFE, Back L, Back R
+  ChannelLayout input_layout = CHANNEL_LAYOUT_5_1_BACK;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_STEREO;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  // Note: 1/sqrt(2) is shown as 0.707.
+  // Note: The Channels enumerator is given by {LEFT = 0, RIGHT, CENTER, LFE,
+  // BACK_LEFT, BACK_RIGHT,...}, hence we can use the enumerator values as
+  // indexes in the matrix when verifying the scaling factors.
+  //
+  //                      Input: 5.1
+  //                      LEFT   RIGHT  CENTER  LFE    BACK_LEFT  BACK_RIGHT
+  // Output: stereo LEFT  1      0      0.707   0.707  0.707      0
+  //                RIGHT 0      1      0.707   0.707  0          0.707
+  //
+  EXPECT_FALSE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  EXPECT_EQ(static_cast<size_t>(input_channels), matrix[LEFT].size());
+  EXPECT_EQ(static_cast<size_t>(input_channels), matrix[RIGHT].size());
+  EXPECT_EQ(1.0f, matrix[LEFT][LEFT]);
+  EXPECT_EQ(1.0f, matrix[RIGHT][RIGHT]);
+  EXPECT_EQ(0.0f, matrix[LEFT][RIGHT]);
+  EXPECT_EQ(0.0f, matrix[RIGHT][LEFT]);
+  EXPECT_EQ(0.0f, matrix[LEFT][BACK_RIGHT]);
+  EXPECT_EQ(0.0f, matrix[RIGHT][BACK_LEFT]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[LEFT][CENTER]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[LEFT][LFE]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[LEFT][BACK_LEFT]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[RIGHT][CENTER]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[RIGHT][LFE]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[RIGHT][BACK_RIGHT]);
+}
+
+TEST(ChannelMixingMatrixTest, FiveOneToSevenOne) {
+  // Front L, Front R, Front C, LFE, Side L, Side R
+  ChannelLayout input_layout = CHANNEL_LAYOUT_5_1;
+  // Front L, Front R, Front C, LFE, Side L, Side R, Back L, Back R
+  ChannelLayout output_layout = CHANNEL_LAYOUT_7_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                        Input: 5.1
+  //                        LEFT   RIGHT  CENTER  LFE    SIDE_LEFT  SIDE_RIGHT
+  // Output: 7.1 LEFT       1      0      0       0      0          0
+  //             RIGHT      0      1      0       0      0          0
+  //             CENTER     0      0      1       0      0          0
+  //             LFE        0      0      0       1      0          0
+  //             SIDE_LEFT  0      0      0       0      1          0
+  //             SIDE_RIGHT 0      0      0       0      0          1
+  //             BACK_LEFT  0      0      0       0      0          0
+  //             BACK_RIGHT 0      0      0       0      0          0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int i = 0; i < output_channels; i++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[i].size());
+    for (int j = 0; j < input_channels; j++) {
+      if (i == j) {
+        EXPECT_EQ(1.0f, matrix[i][j]);
+      } else {
+        EXPECT_EQ(0.0f, matrix[i][j]);
+      }
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, StereoToFiveOne) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_STEREO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_5_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                         Input: Stereo
+  //                         LEFT   RIGHT
+  // Output: 5.1 LEFT        1      0
+  //             RIGHT       0      1
+  //             CENTER      0      0
+  //             LFE         0      0
+  //             SIDE_LEFT   0      0
+  //             SIDE_RIGHT  0      0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int n = 0; n < output_channels; n++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[n].size());
+    if (n == LEFT) {
+      EXPECT_EQ(1.0f, matrix[LEFT][LEFT]);
+      EXPECT_EQ(0.0f, matrix[LEFT][RIGHT]);
+    } else if (n == RIGHT) {
+      EXPECT_EQ(0.0f, matrix[RIGHT][LEFT]);
+      EXPECT_EQ(1.0f, matrix[RIGHT][RIGHT]);
+    } else {
+      EXPECT_EQ(0.0f, matrix[n][LEFT]);
+      EXPECT_EQ(0.0f, matrix[n][RIGHT]);
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, DiscreteToDiscrete) {
+  const struct {
+    int input_channels;
+    int output_channels;
+  } test_case[] = {
+      {2, 2},
+      {2, 5},
+      {5, 2},
+  };
+
+  for (size_t n = 0; n < arraysize(test_case); n++) {
+    int input_channels = test_case[n].input_channels;
+    int output_channels = test_case[n].output_channels;
+    ChannelMixingMatrix matrix_builder(CHANNEL_LAYOUT_DISCRETE, input_channels,
+                                       CHANNEL_LAYOUT_DISCRETE,
+                                       output_channels);
+    std::vector<std::vector<float>> matrix;
+    bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+    EXPECT_TRUE(remapping);
+    EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+    for (int i = 0; i < output_channels; i++) {
+      EXPECT_EQ(static_cast<size_t>(input_channels), matrix[i].size());
+      for (int j = 0; j < input_channels; j++) {
+        if (i == j) {
+          EXPECT_EQ(1.0f, matrix[i][j]);
+        } else {
+          EXPECT_EQ(0.0f, matrix[i][j]);
+        }
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/voip/BUILD.gn b/third_party/libwebrtc/audio/voip/BUILD.gn
new file mode 100644
index 0000000000..e807e2276b
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/BUILD.gn
@@ -0,0 +1,103 @@
+# Copyright(c) 2020 The WebRTC project authors.All Rights Reserved.
+#
+# Use of this source code is governed by a BSD - style license
+# that can be found in the LICENSE file in the root of the source
+# tree.An additional intellectual property rights grant can be found
+# in the file PATENTS.All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_library("voip_core") {
+  sources = [
+    "voip_core.cc",
+    "voip_core.h",
+  ]
+  deps = [
+    ":audio_channel",
+    "..:audio",
+    "../../api:scoped_refptr",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/task_queue",
+    "../../api/voip:voip_api",
+    "../../modules/audio_device:audio_device_api",
+    "../../modules/audio_mixer:audio_mixer_impl",
+    "../../modules/audio_processing:api",
+    "../../rtc_base:criticalsection",
+    "../../rtc_base:logging",
+    "../../rtc_base/synchronization:mutex",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("audio_channel") {
+  sources = [
+    "audio_channel.cc",
+    "audio_channel.h",
+  ]
+  deps = [
+    ":audio_egress",
+    ":audio_ingress",
+    "../../api:transport_api",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/task_queue",
+    "../../api/voip:voip_api",
+    "../../modules/audio_device:audio_device_api",
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:criticalsection",
+    "../../rtc_base:logging",
+    "../../rtc_base:refcount",
+  ]
+}
+
+rtc_library("audio_ingress") {
+  sources = [
+    "audio_ingress.cc",
+    "audio_ingress.h",
+  ]
+  deps = [
+    "..:audio",
+    "../../api:array_view",
+    "../../api:rtp_headers",
+    "../../api:scoped_refptr",
+    "../../api:transport_api",
+    "../../api/audio:audio_mixer_api",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/voip:voip_api",
+    "../../modules/audio_coding",
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:criticalsection",
+    "../../rtc_base:logging",
+    "../../rtc_base:rtc_numerics",
+    "../../rtc_base:safe_minmax",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/synchronization:mutex",
+    "../utility:audio_frame_operations",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("audio_egress") {
+  sources = [
+    "audio_egress.cc",
+    "audio_egress.h",
+  ]
+  deps = [
+    "..:audio",
+    "../../api:sequence_checker",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../api/task_queue",
+    "../../call:audio_sender_interface",
+    "../../modules/audio_coding",
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:logging",
+    "../../rtc_base:rtc_task_queue",
+    "../../rtc_base:timeutils",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:no_unique_address",
+    "../utility:audio_frame_operations",
+  ]
+}
diff --git a/third_party/libwebrtc/audio/voip/audio_channel.cc b/third_party/libwebrtc/audio/voip/audio_channel.cc
new file mode 100644
index 0000000000..a70e33ec38
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/audio_channel.cc
@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/voip/audio_channel.h"
+
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/audio_format.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kRtcpReportIntervalMs = 5000;
+
+}  // namespace
+
+AudioChannel::AudioChannel(
+    Transport* transport,
+    uint32_t local_ssrc,
+    TaskQueueFactory* task_queue_factory,
+    AudioMixer* audio_mixer,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
+    : audio_mixer_(audio_mixer) {
+  RTC_DCHECK(task_queue_factory);
+  RTC_DCHECK(audio_mixer);
+
+  Clock* clock = Clock::GetRealTimeClock();
+  receive_statistics_ = ReceiveStatistics::Create(clock);
+
+  RtpRtcpInterface::Configuration rtp_config;
+  rtp_config.clock = clock;
+  rtp_config.audio = true;
+  rtp_config.receive_statistics = receive_statistics_.get();
+  rtp_config.rtcp_report_interval_ms = kRtcpReportIntervalMs;
+  rtp_config.outgoing_transport = transport;
+  rtp_config.local_media_ssrc = local_ssrc;
+
+  rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
+
+  rtp_rtcp_->SetSendingMediaStatus(false);
+  rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
+
+  ingress_ = std::make_unique<AudioIngress>(rtp_rtcp_.get(), clock,
+                                            receive_statistics_.get(),
+                                            std::move(decoder_factory));
+  egress_ =
+      std::make_unique<AudioEgress>(rtp_rtcp_.get(), clock, task_queue_factory);
+
+  // Set the instance of audio ingress to be part of audio mixer for ADM to
+  // fetch audio samples to play.
+  audio_mixer_->AddSource(ingress_.get());
+}
+
+AudioChannel::~AudioChannel() {
+  if (egress_->IsSending()) {
+    StopSend();
+  }
+  if (ingress_->IsPlaying()) {
+    StopPlay();
+  }
+
+  audio_mixer_->RemoveSource(ingress_.get());
+
+  // TODO(bugs.webrtc.org/11581): unclear if we still need to clear `egress_`
+  // here.
+  egress_.reset();
+  ingress_.reset();
+}
+
+bool AudioChannel::StartSend() {
+  // If encoder has not been set, return false.
+  if (!egress_->StartSend()) {
+    return false;
+  }
+
+  // Start sending with RTP stack if it has not been sending yet.
+  if (!rtp_rtcp_->Sending()) {
+    rtp_rtcp_->SetSendingStatus(true);
+  }
+  return true;
+}
+
+void AudioChannel::StopSend() {
+  egress_->StopSend();
+
+  // Deactivate RTP stack when both sending and receiving are stopped.
+  // SetSendingStatus(false) triggers the transmission of RTCP BYE
+  // message to remote endpoint.
+  if (!ingress_->IsPlaying() && rtp_rtcp_->Sending()) {
+    rtp_rtcp_->SetSendingStatus(false);
+  }
+}
+
+bool AudioChannel::StartPlay() {
+  // If decoders have not been set, return false.
+  if (!ingress_->StartPlay()) {
+    return false;
+  }
+
+  // If RTP stack is not sending then start sending as in recv-only mode, RTCP
+  // receiver report is expected.
+  if (!rtp_rtcp_->Sending()) {
+    rtp_rtcp_->SetSendingStatus(true);
+  }
+  return true;
+}
+
+void AudioChannel::StopPlay() {
+  ingress_->StopPlay();
+
+  // Deactivate RTP stack only when both sending and receiving are stopped.
+  if (!rtp_rtcp_->SendingMedia() && rtp_rtcp_->Sending()) {
+    rtp_rtcp_->SetSendingStatus(false);
+  }
+}
+
+IngressStatistics AudioChannel::GetIngressStatistics() {
+  IngressStatistics ingress_stats;
+  NetworkStatistics stats = ingress_->GetNetworkStatistics();
+  ingress_stats.neteq_stats.total_samples_received = stats.totalSamplesReceived;
+  ingress_stats.neteq_stats.concealed_samples = stats.concealedSamples;
+  ingress_stats.neteq_stats.concealment_events = stats.concealmentEvents;
+  ingress_stats.neteq_stats.jitter_buffer_delay_ms = stats.jitterBufferDelayMs;
+  ingress_stats.neteq_stats.jitter_buffer_emitted_count =
+      stats.jitterBufferEmittedCount;
+  ingress_stats.neteq_stats.jitter_buffer_target_delay_ms =
+      stats.jitterBufferTargetDelayMs;
+  ingress_stats.neteq_stats.inserted_samples_for_deceleration =
+      stats.insertedSamplesForDeceleration;
+  ingress_stats.neteq_stats.removed_samples_for_acceleration =
+      stats.removedSamplesForAcceleration;
+  ingress_stats.neteq_stats.silent_concealed_samples =
+      stats.silentConcealedSamples;
+  ingress_stats.neteq_stats.fec_packets_received = stats.fecPacketsReceived;
+  ingress_stats.neteq_stats.fec_packets_discarded = stats.fecPacketsDiscarded;
+  ingress_stats.neteq_stats.delayed_packet_outage_samples =
+      stats.delayedPacketOutageSamples;
+  ingress_stats.neteq_stats.relative_packet_arrival_delay_ms =
+      stats.relativePacketArrivalDelayMs;
+  ingress_stats.neteq_stats.interruption_count = stats.interruptionCount;
+  ingress_stats.neteq_stats.total_interruption_duration_ms =
+      stats.totalInterruptionDurationMs;
+  ingress_stats.total_duration = ingress_->GetOutputTotalDuration();
+  return ingress_stats;
+}
+
+ChannelStatistics AudioChannel::GetChannelStatistics() {
+  ChannelStatistics channel_stat = ingress_->GetChannelStatistics();
+
+  StreamDataCounters rtp_stats, rtx_stats;
+  rtp_rtcp_->GetSendStreamDataCounters(&rtp_stats, &rtx_stats);
+  channel_stat.bytes_sent =
+      rtp_stats.transmitted.payload_bytes + rtx_stats.transmitted.payload_bytes;
+  channel_stat.packets_sent =
+      rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
+
+  return channel_stat;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/voip/audio_channel.h b/third_party/libwebrtc/audio/voip/audio_channel.h
new file mode 100644
index 0000000000..7338d9faab
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/audio_channel.h
@@ -0,0 +1,131 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_VOIP_AUDIO_CHANNEL_H_
+#define AUDIO_VOIP_AUDIO_CHANNEL_H_
+
+#include <map>
+#include <memory>
+#include <queue>
+#include <utility>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "api/voip/voip_base.h"
+#include "api/voip/voip_statistics.h"
+#include "audio/voip/audio_egress.h"
+#include "audio/voip/audio_ingress.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// AudioChannel represents a single media session and provides APIs over
+// AudioIngress and AudioEgress. Note that a single RTP stack is shared with
+// these two classes as it has both sending and receiving capabilities.
+class AudioChannel : public rtc::RefCountInterface {
+ public:
+  AudioChannel(Transport* transport,
+               uint32_t local_ssrc,
+               TaskQueueFactory* task_queue_factory,
+               AudioMixer* audio_mixer,
+               rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
+  ~AudioChannel() override;
+
+  // Set and get ChannelId that this audio channel belongs for debugging and
+  // logging purpose.
+  void SetId(ChannelId id) { id_ = id; }
+  ChannelId GetId() const { return id_; }
+
+  // APIs to start/stop audio channel on each direction.
+  // StartSend/StartPlay returns false if encoder/decoders
+  // have not been set, respectively.
+  bool StartSend();
+  void StopSend();
+  bool StartPlay();
+  void StopPlay();
+
+  // APIs relayed to AudioEgress.
+  bool IsSendingMedia() const { return egress_->IsSending(); }
+  AudioSender* GetAudioSender() { return egress_.get(); }
+  void SetEncoder(int payload_type,
+                  const SdpAudioFormat& encoder_format,
+                  std::unique_ptr<AudioEncoder> encoder) {
+    egress_->SetEncoder(payload_type, encoder_format, std::move(encoder));
+  }
+  absl::optional<SdpAudioFormat> GetEncoderFormat() const {
+    return egress_->GetEncoderFormat();
+  }
+  void RegisterTelephoneEventType(int rtp_payload_type, int sample_rate_hz) {
+    egress_->RegisterTelephoneEventType(rtp_payload_type, sample_rate_hz);
+  }
+  bool SendTelephoneEvent(int dtmf_event, int duration_ms) {
+    return egress_->SendTelephoneEvent(dtmf_event, duration_ms);
+  }
+  void SetMute(bool enable) { egress_->SetMute(enable); }
+
+  // APIs relayed to AudioIngress.
+  bool IsPlaying() const { return ingress_->IsPlaying(); }
+  void ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
+    ingress_->ReceivedRTPPacket(rtp_packet);
+  }
+  void ReceivedRTCPPacket(rtc::ArrayView<const uint8_t> rtcp_packet) {
+    ingress_->ReceivedRTCPPacket(rtcp_packet);
+  }
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+    ingress_->SetReceiveCodecs(codecs);
+  }
+  IngressStatistics GetIngressStatistics();
+  ChannelStatistics GetChannelStatistics();
+
+  // See comments on the methods used from AudioEgress and AudioIngress.
+  // Conversion to double is following what is done in
+  // DoubleAudioLevelFromIntAudioLevel method in rtc_stats_collector.cc to be
+  // consistent.
+  double GetInputAudioLevel() const {
+    return egress_->GetInputAudioLevel() / 32767.0;
+  }
+  double GetInputTotalEnergy() const { return egress_->GetInputTotalEnergy(); }
+  double GetInputTotalDuration() const {
+    return egress_->GetInputTotalDuration();
+  }
+  double GetOutputAudioLevel() const {
+    return ingress_->GetOutputAudioLevel() / 32767.0;
+  }
+  double GetOutputTotalEnergy() const {
+    return ingress_->GetOutputTotalEnergy();
+  }
+  double GetOutputTotalDuration() const {
+    return ingress_->GetOutputTotalDuration();
+  }
+
+  // Internal API for testing purpose.
+  void SendRTCPReportForTesting(RTCPPacketType type) {
+    int32_t result = rtp_rtcp_->SendRTCP(type);
+    RTC_DCHECK(result == 0);
+  }
+
+ private:
+  // ChannelId that this audio channel belongs for logging purpose.
+  ChannelId id_;
+
+  // Synchronization is handled internally by AudioMixer.
+  AudioMixer* audio_mixer_;
+
+  // Listed in order for safe destruction of AudioChannel object.
+  // Synchronization for these are handled internally.
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
+  std::unique_ptr<AudioIngress> ingress_;
+  std::unique_ptr<AudioEgress> egress_;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_AUDIO_CHANNEL_H_
diff --git a/third_party/libwebrtc/audio/voip/audio_egress.cc b/third_party/libwebrtc/audio/voip/audio_egress.cc
new file mode 100644
index 0000000000..95a1a3351e
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/audio_egress.cc
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/voip/audio_egress.h"
+
+#include <utility>
+#include <vector>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+AudioEgress::AudioEgress(RtpRtcpInterface* rtp_rtcp,
+                         Clock* clock,
+                         TaskQueueFactory* task_queue_factory)
+    : rtp_rtcp_(rtp_rtcp),
+      rtp_sender_audio_(clock, rtp_rtcp_->RtpSender()),
+      audio_coding_(AudioCodingModule::Create()),
+      encoder_queue_(task_queue_factory->CreateTaskQueue(
+          "AudioEncoder",
+          TaskQueueFactory::Priority::NORMAL)) {
+  audio_coding_->RegisterTransportCallback(this);
+}
+
+AudioEgress::~AudioEgress() {
+  audio_coding_->RegisterTransportCallback(nullptr);
+}
+
+bool AudioEgress::IsSending() const {
+  return rtp_rtcp_->SendingMedia();
+}
+
+void AudioEgress::SetEncoder(int payload_type,
+                             const SdpAudioFormat& encoder_format,
+                             std::unique_ptr<AudioEncoder> encoder) {
+  RTC_DCHECK_GE(payload_type, 0);
+  RTC_DCHECK_LE(payload_type, 127);
+
+  SetEncoderFormat(encoder_format);
+
+  // The RTP/RTCP module needs to know the RTP timestamp rate (i.e. clockrate)
+  // as well as some other things, so we collect this info and send it along.
+  rtp_rtcp_->RegisterSendPayloadFrequency(payload_type,
+                                          encoder->RtpTimestampRateHz());
+  rtp_sender_audio_.RegisterAudioPayload("audio", payload_type,
+                                         encoder->RtpTimestampRateHz(),
+                                         encoder->NumChannels(), 0);
+
+  audio_coding_->SetEncoder(std::move(encoder));
+}
+
+bool AudioEgress::StartSend() {
+  if (!GetEncoderFormat()) {
+    RTC_DLOG(LS_WARNING) << "Send codec has not been set yet";
+    return false;
+  }
+  rtp_rtcp_->SetSendingMediaStatus(true);
+  return true;
+}
+
+void AudioEgress::StopSend() {
+  rtp_rtcp_->SetSendingMediaStatus(false);
+}
+
+void AudioEgress::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
+  RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
+  RTC_DCHECK_LE(audio_frame->num_channels_, 8);
+
+  encoder_queue_.PostTask(
+      [this, audio_frame = std::move(audio_frame)]() mutable {
+        RTC_DCHECK_RUN_ON(&encoder_queue_);
+        if (!rtp_rtcp_->SendingMedia()) {
+          return;
+        }
+
+        double duration_seconds =
+            static_cast<double>(audio_frame->samples_per_channel_) /
+            audio_frame->sample_rate_hz_;
+
+        input_audio_level_.ComputeLevel(*audio_frame, duration_seconds);
+
+        AudioFrameOperations::Mute(audio_frame.get(),
+                                   encoder_context_.previously_muted_,
+                                   encoder_context_.mute_);
+        encoder_context_.previously_muted_ = encoder_context_.mute_;
+
+        audio_frame->timestamp_ = encoder_context_.frame_rtp_timestamp_;
+
+        // This call will trigger AudioPacketizationCallback::SendData if
+        // encoding is done and payload is ready for packetization and
+        // transmission. Otherwise, it will return without invoking the
+        // callback.
+        if (audio_coding_->Add10MsData(*audio_frame) < 0) {
+          RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
+          return;
+        }
+
+        encoder_context_.frame_rtp_timestamp_ +=
+            rtc::dchecked_cast<uint32_t>(audio_frame->samples_per_channel_);
+      });
+}
+
+int32_t AudioEgress::SendData(AudioFrameType frame_type,
+                              uint8_t payload_type,
+                              uint32_t timestamp,
+                              const uint8_t* payload_data,
+                              size_t payload_size) {
+  RTC_DCHECK_RUN_ON(&encoder_queue_);
+
+  rtc::ArrayView<const uint8_t> payload(payload_data, payload_size);
+
+  // Currently we don't get a capture time from downstream modules (ADM,
+  // AudioTransportImpl).
+  // TODO(natim@webrtc.org): Integrate once it's ready.
+  constexpr uint32_t kUndefinedCaptureTime = -1;
+
+  // Push data from ACM to RTP/RTCP-module to deliver audio frame for
+  // packetization.
+  if (!rtp_rtcp_->OnSendingRtpFrame(timestamp, kUndefinedCaptureTime,
+                                    payload_type,
+                                    /*force_sender_report=*/false)) {
+    return -1;
+  }
+
+  const uint32_t rtp_timestamp = timestamp + rtp_rtcp_->StartTimestamp();
+
+  // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
+  if (!rtp_sender_audio_.SendAudio({.type = frame_type,
+                                    .payload = payload,
+                                    .payload_id = payload_type,
+                                    .rtp_timestamp = rtp_timestamp})) {
+    RTC_DLOG(LS_ERROR)
+        << "AudioEgress::SendData() failed to send data to RTP/RTCP module";
+    return -1;
+  }
+
+  return 0;
+}
+
+void AudioEgress::RegisterTelephoneEventType(int rtp_payload_type,
+                                             int sample_rate_hz) {
+  RTC_DCHECK_GE(rtp_payload_type, 0);
+  RTC_DCHECK_LE(rtp_payload_type, 127);
+
+  rtp_rtcp_->RegisterSendPayloadFrequency(rtp_payload_type, sample_rate_hz);
+  rtp_sender_audio_.RegisterAudioPayload("telephone-event", rtp_payload_type,
+                                         sample_rate_hz, 0, 0);
+}
+
+bool AudioEgress::SendTelephoneEvent(int dtmf_event, int duration_ms) {
+  RTC_DCHECK_GE(dtmf_event, 0);
+  RTC_DCHECK_LE(dtmf_event, 255);
+  RTC_DCHECK_GE(duration_ms, 0);
+  RTC_DCHECK_LE(duration_ms, 65535);
+
+  if (!IsSending()) {
+    return false;
+  }
+
+  constexpr int kTelephoneEventAttenuationdB = 10;
+
+  if (rtp_sender_audio_.SendTelephoneEvent(dtmf_event, duration_ms,
+                                           kTelephoneEventAttenuationdB) != 0) {
+    RTC_DLOG(LS_ERROR) << "SendTelephoneEvent() failed to send event";
+    return false;
+  }
+  return true;
+}
+
+void AudioEgress::SetMute(bool mute) {
+  encoder_queue_.PostTask([this, mute] {
+    RTC_DCHECK_RUN_ON(&encoder_queue_);
+    encoder_context_.mute_ = mute;
+  });
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/voip/audio_egress.h b/third_party/libwebrtc/audio/voip/audio_egress.h
new file mode 100644
index 0000000000..989e5bda59
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/audio_egress.h
@@ -0,0 +1,158 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_VOIP_AUDIO_EGRESS_H_
+#define AUDIO_VOIP_AUDIO_EGRESS_H_
+
+#include <memory>
+#include <string>
+
+#include "api/audio_codecs/audio_format.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "audio/audio_level.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "call/audio_sender.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+// AudioEgress receives input samples from AudioDeviceModule via
+// AudioTransportImpl through AudioSender interface. Once it encodes the sample
+// via selected encoder through AudioPacketizationCallback interface, the
+// encoded payload will be packetized by the RTP stack, resulting in ready to
+// send RTP packet to remote endpoint.
+//
+// TaskQueue is used to encode and send RTP asynchrounously as some OS platform
+// uses the same thread for both audio input and output sample deliveries which
+// can affect audio quality.
+//
+// Note that this class is originally based on ChannelSend in
+// audio/channel_send.cc with non-audio related logic trimmed as aimed for
+// smaller footprint.
+class AudioEgress : public AudioSender, public AudioPacketizationCallback {
+ public:
+  AudioEgress(RtpRtcpInterface* rtp_rtcp,
+              Clock* clock,
+              TaskQueueFactory* task_queue_factory);
+  ~AudioEgress() override;
+
+  // Set the encoder format and payload type for AudioCodingModule.
+  // It's possible to change the encoder type during its active usage.
+  // `payload_type` must be the type that is negotiated with peer through
+  // offer/answer.
+  void SetEncoder(int payload_type,
+                  const SdpAudioFormat& encoder_format,
+                  std::unique_ptr<AudioEncoder> encoder);
+
+  // Start or stop sending operation of AudioEgress. This will start/stop
+  // the RTP stack also causes encoder queue thread to start/stop
+  // processing input audio samples. StartSend will return false if
+  // a send codec has not been set.
+  bool StartSend();
+  void StopSend();
+
+  // Query the state of the RTP stack. This returns true if StartSend()
+  // called and false if StopSend() is called.
+  bool IsSending() const;
+
+  // Enable or disable Mute state.
+  void SetMute(bool mute);
+
+  // Retrieve current encoder format info. This returns encoder format set
+  // by SetEncoder() and if encoder is not set, this will return nullopt.
+  absl::optional<SdpAudioFormat> GetEncoderFormat() const {
+    MutexLock lock(&lock_);
+    return encoder_format_;
+  }
+
+  // Register the payload type and sample rate for DTMF (RFC 4733) payload.
+  void RegisterTelephoneEventType(int rtp_payload_type, int sample_rate_hz);
+
+  // Send DTMF named event as specified by
+  // https://tools.ietf.org/html/rfc4733#section-3.2
+  // `duration_ms` specifies the duration of DTMF packets that will be emitted
+  // in place of real RTP packets instead.
+  // This will return true when requested dtmf event is successfully scheduled
+  // otherwise false when the dtmf queue reached maximum of 20 events.
+  bool SendTelephoneEvent(int dtmf_event, int duration_ms);
+
+  // See comments on LevelFullRange, TotalEnergy, TotalDuration from
+  // audio/audio_level.h.
+  int GetInputAudioLevel() const { return input_audio_level_.LevelFullRange(); }
+  double GetInputTotalEnergy() const {
+    return input_audio_level_.TotalEnergy();
+  }
+  double GetInputTotalDuration() const {
+    return input_audio_level_.TotalDuration();
+  }
+
+  // Implementation of AudioSender interface.
+  void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override;
+
+  // Implementation of AudioPacketizationCallback interface.
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_size) override;
+
+ private:
+  void SetEncoderFormat(const SdpAudioFormat& encoder_format) {
+    MutexLock lock(&lock_);
+    encoder_format_ = encoder_format;
+  }
+
+  mutable Mutex lock_;
+
+  // Current encoder format selected by caller.
+  absl::optional<SdpAudioFormat> encoder_format_ RTC_GUARDED_BY(lock_);
+
+  // Synchronization is handled internally by RtpRtcp.
+  RtpRtcpInterface* const rtp_rtcp_;
+
+  // Synchronization is handled internally by RTPSenderAudio.
+  RTPSenderAudio rtp_sender_audio_;
+
+  // Synchronization is handled internally by AudioCodingModule.
+  const std::unique_ptr<AudioCodingModule> audio_coding_;
+
+  // Synchronization is handled internally by voe::AudioLevel.
+  voe::AudioLevel input_audio_level_;
+
+  // Struct that holds all variables used by encoder task queue.
+  struct EncoderContext {
+    // Offset used to mark rtp timestamp in sample rate unit in
+    // newly received audio frame from AudioTransport.
+    uint32_t frame_rtp_timestamp_ = 0;
+
+    // Flag to track mute state from caller. `previously_muted_` is used to
+    // track previous state as part of input to AudioFrameOperations::Mute
+    // to implement fading effect when (un)mute is invoked.
+    bool mute_ = false;
+    bool previously_muted_ = false;
+  };
+
+  EncoderContext encoder_context_ RTC_GUARDED_BY(encoder_queue_);
+
+  // Defined last to ensure that there are no running tasks when the other
+  // members are destroyed.
+  rtc::TaskQueue encoder_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_AUDIO_EGRESS_H_
diff --git a/third_party/libwebrtc/audio/voip/audio_ingress.cc b/third_party/libwebrtc/audio/voip/audio_ingress.cc
new file mode 100644
index 0000000000..80f21152c0
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/audio_ingress.cc
@@ -0,0 +1,290 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/voip/audio_ingress.h"
+
+#include <algorithm>
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/audio_format.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+
+acm2::AcmReceiver::Config CreateAcmConfig(
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) {
+  acm2::AcmReceiver::Config acm_config;
+  acm_config.neteq_config.enable_muted_state = true;
+  acm_config.decoder_factory = decoder_factory;
+  return acm_config;
+}
+
+}  // namespace
+
+AudioIngress::AudioIngress(
+    RtpRtcpInterface* rtp_rtcp,
+    Clock* clock,
+    ReceiveStatistics* receive_statistics,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
+    : playing_(false),
+      remote_ssrc_(0),
+      first_rtp_timestamp_(-1),
+      rtp_receive_statistics_(receive_statistics),
+      rtp_rtcp_(rtp_rtcp),
+      acm_receiver_(CreateAcmConfig(decoder_factory)),
+      ntp_estimator_(clock) {}
+
+AudioIngress::~AudioIngress() = default;
+
+AudioMixer::Source::AudioFrameInfo AudioIngress::GetAudioFrameWithInfo(
+    int sampling_rate,
+    AudioFrame* audio_frame) {
+  audio_frame->sample_rate_hz_ = sampling_rate;
+
+  // Get 10ms raw PCM data from the ACM.
+  bool muted = false;
+  if (acm_receiver_.GetAudio(sampling_rate, audio_frame, &muted) == -1) {
+    RTC_DLOG(LS_ERROR) << "GetAudio() failed!";
+    // In all likelihood, the audio in this frame is garbage. We return an
+    // error so that the audio mixer module doesn't add it to the mix. As
+    // a result, it won't be played out and the actions skipped here are
+    // irrelevant.
+    return AudioMixer::Source::AudioFrameInfo::kError;
+  }
+
+  if (muted) {
+    AudioFrameOperations::Mute(audio_frame);
+  }
+
+  // Measure audio level.
+  constexpr double kAudioSampleDurationSeconds = 0.01;
+  output_audio_level_.ComputeLevel(*audio_frame, kAudioSampleDurationSeconds);
+
+  // If caller invoked StopPlay(), then mute the frame.
+  if (!playing_) {
+    AudioFrameOperations::Mute(audio_frame);
+    muted = true;
+  }
+
+  // Set first rtp timestamp with first audio frame with valid timestamp.
+  if (first_rtp_timestamp_ < 0 && audio_frame->timestamp_ != 0) {
+    first_rtp_timestamp_ = audio_frame->timestamp_;
+  }
+
+  if (first_rtp_timestamp_ >= 0) {
+    // Compute elapsed and NTP times.
+    int64_t unwrap_timestamp;
+    {
+      MutexLock lock(&lock_);
+      unwrap_timestamp =
+          timestamp_wrap_handler_.Unwrap(audio_frame->timestamp_);
+      audio_frame->ntp_time_ms_ =
+          ntp_estimator_.Estimate(audio_frame->timestamp_);
+    }
+    // For clock rate, default to the playout sampling rate if we haven't
+    // received any packets yet.
+    absl::optional<std::pair<int, SdpAudioFormat>> decoder =
+        acm_receiver_.LastDecoder();
+    int clock_rate = decoder ? decoder->second.clockrate_hz
+                             : acm_receiver_.last_output_sample_rate_hz();
+    RTC_DCHECK_GT(clock_rate, 0);
+    audio_frame->elapsed_time_ms_ =
+        (unwrap_timestamp - first_rtp_timestamp_) / (clock_rate / 1000);
+  }
+
+  return muted ? AudioMixer::Source::AudioFrameInfo::kMuted
+               : AudioMixer::Source::AudioFrameInfo::kNormal;
+}
+
+bool AudioIngress::StartPlay() {
+  {
+    MutexLock lock(&lock_);
+    if (receive_codec_info_.empty()) {
+      RTC_DLOG(LS_WARNING) << "Receive codecs have not been set yet";
+      return false;
+    }
+  }
+  playing_ = true;
+  return true;
+}
+
+void AudioIngress::SetReceiveCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  {
+    MutexLock lock(&lock_);
+    for (const auto& kv : codecs) {
+      receive_codec_info_[kv.first] = kv.second.clockrate_hz;
+    }
+  }
+  acm_receiver_.SetCodecs(codecs);
+}
+
+void AudioIngress::ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet) {
+  RtpPacketReceived rtp_packet_received;
+  rtp_packet_received.Parse(rtp_packet.data(), rtp_packet.size());
+
+  // Set payload type's sampling rate before we feed it into ReceiveStatistics.
+  {
+    MutexLock lock(&lock_);
+    const auto& it =
+        receive_codec_info_.find(rtp_packet_received.PayloadType());
+    // If sampling rate info is not available in our received codec set, it
+    // would mean that remote media endpoint is sending incorrect payload id
+    // which can't be processed correctly especially on payload type id in
+    // dynamic range.
+    if (it == receive_codec_info_.end()) {
+      RTC_DLOG(LS_WARNING) << "Unexpected payload id received: "
+                           << rtp_packet_received.PayloadType();
+      return;
+    }
+    rtp_packet_received.set_payload_type_frequency(it->second);
+  }
+
+  // Track current remote SSRC.
+  if (rtp_packet_received.Ssrc() != remote_ssrc_) {
+    rtp_rtcp_->SetRemoteSSRC(rtp_packet_received.Ssrc());
+    remote_ssrc_.store(rtp_packet_received.Ssrc());
+  }
+
+  rtp_receive_statistics_->OnRtpPacket(rtp_packet_received);
+
+  RTPHeader header;
+  rtp_packet_received.GetHeader(&header);
+
+  size_t packet_length = rtp_packet_received.size();
+  if (packet_length < header.headerLength ||
+      (packet_length - header.headerLength) < header.paddingLength) {
+    RTC_DLOG(LS_ERROR) << "Packet length(" << packet_length << ") header("
+                       << header.headerLength << ") padding("
+                       << header.paddingLength << ")";
+    return;
+  }
+
+  const uint8_t* payload = rtp_packet_received.data() + header.headerLength;
+  size_t payload_length = packet_length - header.headerLength;
+  size_t payload_data_length = payload_length - header.paddingLength;
+  auto data_view = rtc::ArrayView<const uint8_t>(payload, payload_data_length);
+
+  // Push the incoming payload (parsed and ready for decoding) into the ACM.
+  if (acm_receiver_.InsertPacket(header, data_view) != 0) {
+    RTC_DLOG(LS_ERROR) << "AudioIngress::ReceivedRTPPacket() unable to "
+                          "push data to the ACM";
+  }
+}
+
+void AudioIngress::ReceivedRTCPPacket(
+    rtc::ArrayView<const uint8_t> rtcp_packet) {
+  rtcp::CommonHeader rtcp_header;
+  if (rtcp_header.Parse(rtcp_packet.data(), rtcp_packet.size()) &&
+      (rtcp_header.type() == rtcp::SenderReport::kPacketType ||
+       rtcp_header.type() == rtcp::ReceiverReport::kPacketType)) {
+    RTC_DCHECK_GE(rtcp_packet.size(), 8);
+
+    uint32_t sender_ssrc =
+        ByteReader<uint32_t>::ReadBigEndian(rtcp_packet.data() + 4);
+
+    // If we don't have remote ssrc at this point, it's likely that remote
+    // endpoint is receive-only or it could have restarted the media.
+    if (sender_ssrc != remote_ssrc_) {
+      rtp_rtcp_->SetRemoteSSRC(sender_ssrc);
+      remote_ssrc_.store(sender_ssrc);
+    }
+  }
+
+  // Deliver RTCP packet to RTP/RTCP module for parsing and processing.
+  rtp_rtcp_->IncomingRtcpPacket(rtcp_packet);
+
+  absl::optional<TimeDelta> rtt = rtp_rtcp_->LastRtt();
+  if (!rtt.has_value()) {
+    // Waiting for valid RTT.
+    return;
+  }
+
+  absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+      rtp_rtcp_->GetSenderReportStats();
+  if (!last_sr.has_value()) {
+    // Waiting for RTCP.
+    return;
+  }
+
+  {
+    MutexLock lock(&lock_);
+    ntp_estimator_.UpdateRtcpTimestamp(*rtt, last_sr->last_remote_timestamp,
+                                       last_sr->last_remote_rtp_timestamp);
+  }
+}
+
+ChannelStatistics AudioIngress::GetChannelStatistics() {
+  ChannelStatistics channel_stats;
+
+  // Get clockrate for current decoder ahead of jitter calculation.
+  uint32_t clockrate_hz = 0;
+  absl::optional<std::pair<int, SdpAudioFormat>> decoder =
+      acm_receiver_.LastDecoder();
+  if (decoder) {
+    clockrate_hz = decoder->second.clockrate_hz;
+  }
+
+  StreamStatistician* statistician =
+      rtp_receive_statistics_->GetStatistician(remote_ssrc_);
+  if (statistician) {
+    RtpReceiveStats stats = statistician->GetStats();
+    channel_stats.packets_lost = stats.packets_lost;
+    channel_stats.packets_received = stats.packet_counter.packets;
+    channel_stats.bytes_received = stats.packet_counter.payload_bytes;
+    channel_stats.remote_ssrc = remote_ssrc_;
+    if (clockrate_hz > 0) {
+      channel_stats.jitter = static_cast<double>(stats.jitter) / clockrate_hz;
+    }
+  }
+
+  // Get RTCP report using remote SSRC.
+  const std::vector<ReportBlockData>& report_data =
+      rtp_rtcp_->GetLatestReportBlockData();
+  for (const ReportBlockData& rtcp_report : report_data) {
+    if (rtp_rtcp_->SSRC() != rtcp_report.source_ssrc() ||
+        remote_ssrc_ != rtcp_report.sender_ssrc()) {
+      continue;
+    }
+    RemoteRtcpStatistics remote_stat;
+    remote_stat.packets_lost = rtcp_report.cumulative_lost();
+    remote_stat.fraction_lost = rtcp_report.fraction_lost();
+    if (clockrate_hz > 0) {
+      remote_stat.jitter = rtcp_report.jitter(clockrate_hz).seconds<double>();
+    }
+    if (rtcp_report.has_rtt()) {
+      remote_stat.round_trip_time = rtcp_report.last_rtt().seconds<double>();
+    }
+    remote_stat.last_report_received_timestamp_ms =
+        rtcp_report.report_block_timestamp_utc().ms();
+    channel_stats.remote_rtcp = remote_stat;
+
+    // Receive only channel won't send any RTP packets.
+    if (!channel_stats.remote_ssrc.has_value()) {
+      channel_stats.remote_ssrc = remote_ssrc_;
+    }
+    break;
+  }
+
+  return channel_stats;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/voip/audio_ingress.h b/third_party/libwebrtc/audio/voip/audio_ingress.h
new file mode 100644
index 0000000000..11bde7ce28
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/audio_ingress.h
@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_VOIP_AUDIO_INGRESS_H_
+#define AUDIO_VOIP_AUDIO_INGRESS_H_
+
+#include <algorithm>
+#include <atomic>
+#include <map>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/audio_mixer.h"
+#include "api/rtp_headers.h"
+#include "api/scoped_refptr.h"
+#include "api/voip/voip_statistics.h"
+#include "audio/audio_level.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// AudioIngress handles incoming RTP/RTCP packets from the remote
+// media endpoint. Received RTP packets are injected into AcmReceiver and
+// when audio output thread requests for audio samples to play through system
+// output such as speaker device, AudioIngress provides the samples via its
+// implementation on AudioMixer::Source interface.
+//
+// Note that this class is originally based on ChannelReceive in
+// audio/channel_receive.cc with non-audio related logic trimmed as aimed for
+// smaller footprint.
+class AudioIngress : public AudioMixer::Source {
+ public:
+  AudioIngress(RtpRtcpInterface* rtp_rtcp,
+               Clock* clock,
+               ReceiveStatistics* receive_statistics,
+               rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
+  ~AudioIngress() override;
+
+  // Start or stop receiving operation of AudioIngress.
+  bool StartPlay();
+  void StopPlay() {
+    playing_ = false;
+    output_audio_level_.ResetLevelFullRange();
+  }
+
+  // Query the state of the AudioIngress.
+  bool IsPlaying() const { return playing_; }
+
+  // Set the decoder formats and payload type for AcmReceiver where the
+  // key type (int) of the map is the payload type of SdpAudioFormat.
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
+
+  // APIs to handle received RTP/RTCP packets from caller.
+  void ReceivedRTPPacket(rtc::ArrayView<const uint8_t> rtp_packet);
+  void ReceivedRTCPPacket(rtc::ArrayView<const uint8_t> rtcp_packet);
+
+  // See comments on LevelFullRange, TotalEnergy, TotalDuration from
+  // audio/audio_level.h.
+  int GetOutputAudioLevel() const {
+    return output_audio_level_.LevelFullRange();
+  }
+  double GetOutputTotalEnergy() { return output_audio_level_.TotalEnergy(); }
+  double GetOutputTotalDuration() {
+    return output_audio_level_.TotalDuration();
+  }
+
+  NetworkStatistics GetNetworkStatistics() const {
+    NetworkStatistics stats;
+    acm_receiver_.GetNetworkStatistics(&stats,
+                                       /*get_and_clear_legacy_stats=*/false);
+    return stats;
+  }
+
+  ChannelStatistics GetChannelStatistics();
+
+  // Implementation of AudioMixer::Source interface.
+  AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
+      int sampling_rate,
+      AudioFrame* audio_frame) override;
+  int Ssrc() const override {
+    return rtc::dchecked_cast<int>(remote_ssrc_.load());
+  }
+  int PreferredSampleRate() const override {
+    // If we haven't received any RTP packet from remote and thus
+    // last_packet_sampling_rate is not available then use NetEq's sampling
+    // rate as that would be what would be used for audio output sample.
+    return std::max(acm_receiver_.last_packet_sample_rate_hz().value_or(0),
+                    acm_receiver_.last_output_sample_rate_hz());
+  }
+
+ private:
+  // Indicates AudioIngress status as caller invokes Start/StopPlaying.
+  // If not playing, incoming RTP data processing is skipped, thus
+  // producing no data to output device.
+  std::atomic<bool> playing_;
+
+  // Currently active remote ssrc from remote media endpoint.
+  std::atomic<uint32_t> remote_ssrc_;
+
+  // The first rtp timestamp of the output audio frame that is used to
+  // calculate elasped time for subsequent audio frames.
+  std::atomic<int64_t> first_rtp_timestamp_;
+
+  // Synchronizaton is handled internally by ReceiveStatistics.
+  ReceiveStatistics* const rtp_receive_statistics_;
+
+  // Synchronizaton is handled internally by RtpRtcpInterface.
+  RtpRtcpInterface* const rtp_rtcp_;
+
+  // Synchronizaton is handled internally by acm2::AcmReceiver.
+  acm2::AcmReceiver acm_receiver_;
+
+  // Synchronizaton is handled internally by voe::AudioLevel.
+  voe::AudioLevel output_audio_level_;
+
+  Mutex lock_;
+
+  RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(lock_);
+
+  // For receiving RTP statistics, this tracks the sampling rate value
+  // per payload type set when caller set via SetReceiveCodecs.
+  std::map<int, int> receive_codec_info_ RTC_GUARDED_BY(lock_);
+
+  RtpTimestampUnwrapper timestamp_wrap_handler_ RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_AUDIO_INGRESS_H_
diff --git a/third_party/libwebrtc/audio/voip/test/BUILD.gn b/third_party/libwebrtc/audio/voip/test/BUILD.gn
new file mode 100644
index 0000000000..00e9bee622
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/test/BUILD.gn
@@ -0,0 +1,107 @@
+# Copyright(c) 2020 The WebRTC project authors.All Rights Reserved.
+#
+# Use of this source code is governed by a BSD - style license
+# that can be found in the LICENSE file in the root of the source
+# tree.An additional intellectual property rights grant can be found
+# in the file PATENTS.All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+if (rtc_include_tests) {
+  rtc_source_set("mock_task_queue") {
+    testonly = true
+    visibility = [ "*" ]
+    sources = [ "mock_task_queue.h" ]
+    deps = [
+      "../../../api/task_queue:task_queue",
+      "../../../api/task_queue/test:mock_task_queue_base",
+      "../../../test:test_support",
+    ]
+  }
+
+  if (!build_with_chromium) {
+    rtc_library("voip_core_unittests") {
+      testonly = true
+      sources = [ "voip_core_unittest.cc" ]
+      deps = [
+        "..:voip_core",
+        "../../../api/audio_codecs:builtin_audio_decoder_factory",
+        "../../../api/audio_codecs:builtin_audio_encoder_factory",
+        "../../../api/task_queue:default_task_queue_factory",
+        "../../../modules/audio_device:mock_audio_device",
+        "../../../modules/audio_processing:mocks",
+        "../../../test:audio_codec_mocks",
+        "../../../test:mock_transport",
+        "../../../test:run_loop",
+        "../../../test:test_support",
+      ]
+    }
+  }
+
+  rtc_library("audio_channel_unittests") {
+    testonly = true
+    sources = [ "audio_channel_unittest.cc" ]
+    deps = [
+      ":mock_task_queue",
+      "..:audio_channel",
+      "../../../api:transport_api",
+      "../../../api/audio_codecs:builtin_audio_decoder_factory",
+      "../../../api/audio_codecs:builtin_audio_encoder_factory",
+      "../../../api/task_queue:task_queue",
+      "../../../modules/audio_mixer:audio_mixer_impl",
+      "../../../modules/audio_mixer:audio_mixer_test_utils",
+      "../../../modules/rtp_rtcp:rtp_rtcp",
+      "../../../modules/rtp_rtcp:rtp_rtcp_format",
+      "../../../rtc_base:logging",
+      "../../../test:mock_transport",
+      "../../../test:test_support",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ]
+  }
+
+  rtc_library("audio_ingress_unittests") {
+    testonly = true
+    sources = [ "audio_ingress_unittest.cc" ]
+    deps = [
+      "..:audio_egress",
+      "..:audio_ingress",
+      "../../../api:transport_api",
+      "../../../api/audio_codecs:builtin_audio_decoder_factory",
+      "../../../api/audio_codecs:builtin_audio_encoder_factory",
+      "../../../api/task_queue:default_task_queue_factory",
+      "../../../api/units:time_delta",
+      "../../../api/units:timestamp",
+      "../../../modules/audio_mixer:audio_mixer_test_utils",
+      "../../../modules/rtp_rtcp:rtp_rtcp",
+      "../../../rtc_base:logging",
+      "../../../rtc_base:rtc_event",
+      "../../../test:mock_transport",
+      "../../../test:run_loop",
+      "../../../test:test_support",
+      "../../../test/time_controller:time_controller",
+    ]
+  }
+
+  rtc_library("audio_egress_unittests") {
+    testonly = true
+    sources = [ "audio_egress_unittest.cc" ]
+    deps = [
+      "..:audio_egress",
+      "../../../api:transport_api",
+      "../../../api/audio_codecs:builtin_audio_encoder_factory",
+      "../../../api/task_queue:default_task_queue_factory",
+      "../../../api/units:time_delta",
+      "../../../api/units:timestamp",
+      "../../../modules/audio_mixer:audio_mixer_test_utils",
+      "../../../modules/rtp_rtcp:rtp_rtcp",
+      "../../../modules/rtp_rtcp:rtp_rtcp_format",
+      "../../../rtc_base:logging",
+      "../../../rtc_base:rtc_event",
+      "../../../test:mock_transport",
+      "../../../test:run_loop",
+      "../../../test:test_support",
+      "../../../test/time_controller:time_controller",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/audio/voip/test/audio_channel_unittest.cc b/third_party/libwebrtc/audio/voip/test/audio_channel_unittest.cc
new file mode 100644
index 0000000000..0c8312b738
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/test/audio_channel_unittest.cc
@@ -0,0 +1,355 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/voip/audio_channel.h"
+
+#include "absl/functional/any_invocable.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/call/transport.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "audio/voip/test/mock_task_queue.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_mixer/sine_wave_generator.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/logging.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_transport.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Invoke;
+using ::testing::NiceMock;
+using ::testing::Return;
+using ::testing::Unused;
+using ::testing::WithArg;
+
+constexpr uint64_t kStartTime = 123456789;
+constexpr uint32_t kLocalSsrc = 0xdeadc0de;
+constexpr int16_t kAudioLevel = 3004;  // used for sine wave level
+constexpr int kPcmuPayload = 0;
+
+class AudioChannelTest : public ::testing::Test {
+ public:
+  const SdpAudioFormat kPcmuFormat = {"pcmu", 8000, 1};
+
+  AudioChannelTest()
+      : fake_clock_(kStartTime), wave_generator_(1000.0, kAudioLevel) {
+    task_queue_factory_ = std::make_unique<MockTaskQueueFactory>(&task_queue_);
+    audio_mixer_ = AudioMixerImpl::Create();
+    encoder_factory_ = CreateBuiltinAudioEncoderFactory();
+    decoder_factory_ = CreateBuiltinAudioDecoderFactory();
+
+    // By default, run the queued task immediately.
+    ON_CALL(task_queue_, PostTaskImpl)
+        .WillByDefault(WithArg<0>(
+            [](absl::AnyInvocable<void() &&> task) { std::move(task)(); }));
+  }
+
+  void SetUp() override { audio_channel_ = CreateAudioChannel(kLocalSsrc); }
+
+  void TearDown() override { audio_channel_ = nullptr; }
+
+  rtc::scoped_refptr<AudioChannel> CreateAudioChannel(uint32_t ssrc) {
+    // Use same audio mixer here for simplicity sake as we are not checking
+    // audio activity of RTP in our testcases. If we need to do test on audio
+    // signal activity then we need to assign audio mixer for each channel.
+    // Also this uses the same transport object for different audio channel to
+    // simplify network routing logic.
+    rtc::scoped_refptr<AudioChannel> audio_channel =
+        rtc::make_ref_counted<AudioChannel>(
+            &transport_, ssrc, task_queue_factory_.get(), audio_mixer_.get(),
+            decoder_factory_);
+    audio_channel->SetEncoder(kPcmuPayload, kPcmuFormat,
+                              encoder_factory_->MakeAudioEncoder(
+                                  kPcmuPayload, kPcmuFormat, absl::nullopt));
+    audio_channel->SetReceiveCodecs({{kPcmuPayload, kPcmuFormat}});
+    audio_channel->StartSend();
+    audio_channel->StartPlay();
+    return audio_channel;
+  }
+
+  std::unique_ptr<AudioFrame> GetAudioFrame(int order) {
+    auto frame = std::make_unique<AudioFrame>();
+    frame->sample_rate_hz_ = kPcmuFormat.clockrate_hz;
+    frame->samples_per_channel_ = kPcmuFormat.clockrate_hz / 100;  // 10 ms.
+    frame->num_channels_ = kPcmuFormat.num_channels;
+    frame->timestamp_ = frame->samples_per_channel_ * order;
+    wave_generator_.GenerateNextFrame(frame.get());
+    return frame;
+  }
+
+  SimulatedClock fake_clock_;
+  SineWaveGenerator wave_generator_;
+  NiceMock<MockTransport> transport_;
+  NiceMock<MockTaskQueue> task_queue_;
+  std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+  rtc::scoped_refptr<AudioMixer> audio_mixer_;
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
+  rtc::scoped_refptr<AudioChannel> audio_channel_;
+};
+
+// Validate RTP packet generation by feeding audio frames with sine wave.
+// Resulted RTP packet is looped back into AudioChannel and gets decoded into
+// audio frame to see if it has some signal to indicate its validity.
+TEST_F(AudioChannelTest, PlayRtpByLocalLoop) {
+  auto loop_rtp = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    audio_channel_->ReceivedRTPPacket(packet);
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillOnce(Invoke(loop_rtp));
+
+  auto audio_sender = audio_channel_->GetAudioSender();
+  audio_sender->SendAudioData(GetAudioFrame(0));
+  audio_sender->SendAudioData(GetAudioFrame(1));
+
+  AudioFrame empty_frame, audio_frame;
+  empty_frame.Mute();
+  empty_frame.mutable_data();  // This will zero out the data.
+  audio_frame.CopyFrom(empty_frame);
+  audio_mixer_->Mix(/*number_of_channels*/ 1, &audio_frame);
+
+  // We expect now audio frame to pick up something.
+  EXPECT_NE(memcmp(empty_frame.data(), audio_frame.data(),
+                   AudioFrame::kMaxDataSizeBytes),
+            0);
+}
+
+// Validate assigned local SSRC is resulted in RTP packet.
+TEST_F(AudioChannelTest, VerifyLocalSsrcAsAssigned) {
+  RtpPacketReceived rtp;
+  auto loop_rtp = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    rtp.Parse(packet);
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillOnce(Invoke(loop_rtp));
+
+  auto audio_sender = audio_channel_->GetAudioSender();
+  audio_sender->SendAudioData(GetAudioFrame(0));
+  audio_sender->SendAudioData(GetAudioFrame(1));
+
+  EXPECT_EQ(rtp.Ssrc(), kLocalSsrc);
+}
+
+// Check metrics after processing an RTP packet.
+TEST_F(AudioChannelTest, TestIngressStatistics) {
+  auto loop_rtp = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    audio_channel_->ReceivedRTPPacket(packet);
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(loop_rtp));
+
+  auto audio_sender = audio_channel_->GetAudioSender();
+  audio_sender->SendAudioData(GetAudioFrame(0));
+  audio_sender->SendAudioData(GetAudioFrame(1));
+
+  AudioFrame audio_frame;
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+
+  absl::optional<IngressStatistics> ingress_stats =
+      audio_channel_->GetIngressStatistics();
+  EXPECT_TRUE(ingress_stats);
+  EXPECT_EQ(ingress_stats->neteq_stats.total_samples_received, 160ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.concealed_samples, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.concealment_events, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.inserted_samples_for_deceleration, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.removed_samples_for_acceleration, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.silent_concealed_samples, 0ULL);
+  // To extract the jitter buffer length in millisecond, jitter_buffer_delay_ms
+  // needs to be divided by jitter_buffer_emitted_count (number of samples).
+  EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_delay_ms, 1600ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_emitted_count, 160ULL);
+  EXPECT_GT(ingress_stats->neteq_stats.jitter_buffer_target_delay_ms, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.interruption_count, 0);
+  EXPECT_EQ(ingress_stats->neteq_stats.total_interruption_duration_ms, 0);
+  EXPECT_DOUBLE_EQ(ingress_stats->total_duration, 0.02);
+
+  // Now without any RTP pending in jitter buffer pull more.
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+
+  // Send another RTP packet to intentionally break PLC.
+  audio_sender->SendAudioData(GetAudioFrame(2));
+  audio_sender->SendAudioData(GetAudioFrame(3));
+
+  ingress_stats = audio_channel_->GetIngressStatistics();
+  EXPECT_TRUE(ingress_stats);
+  EXPECT_EQ(ingress_stats->neteq_stats.total_samples_received, 320ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.concealed_samples, 168ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.concealment_events, 1ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.inserted_samples_for_deceleration, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.removed_samples_for_acceleration, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.silent_concealed_samples, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_delay_ms, 1600ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_emitted_count, 160ULL);
+  EXPECT_GT(ingress_stats->neteq_stats.jitter_buffer_target_delay_ms, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.interruption_count, 0);
+  EXPECT_EQ(ingress_stats->neteq_stats.total_interruption_duration_ms, 0);
+  EXPECT_DOUBLE_EQ(ingress_stats->total_duration, 0.04);
+
+  // Pull the last RTP packet.
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+
+  ingress_stats = audio_channel_->GetIngressStatistics();
+  EXPECT_TRUE(ingress_stats);
+  EXPECT_EQ(ingress_stats->neteq_stats.total_samples_received, 480ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.concealed_samples, 168ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.concealment_events, 1ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.inserted_samples_for_deceleration, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.removed_samples_for_acceleration, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.silent_concealed_samples, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_delay_ms, 3200ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.jitter_buffer_emitted_count, 320ULL);
+  EXPECT_GT(ingress_stats->neteq_stats.jitter_buffer_target_delay_ms, 0ULL);
+  EXPECT_EQ(ingress_stats->neteq_stats.interruption_count, 0);
+  EXPECT_EQ(ingress_stats->neteq_stats.total_interruption_duration_ms, 0);
+  EXPECT_DOUBLE_EQ(ingress_stats->total_duration, 0.06);
+}
+
+// Check ChannelStatistics metric after processing RTP and RTCP packets.
+TEST_F(AudioChannelTest, TestChannelStatistics) {
+  auto loop_rtp = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    audio_channel_->ReceivedRTPPacket(packet);
+    return true;
+  };
+  auto loop_rtcp = [&](rtc::ArrayView<const uint8_t> packet) {
+    audio_channel_->ReceivedRTCPPacket(packet);
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(loop_rtp));
+  EXPECT_CALL(transport_, SendRtcp).WillRepeatedly(Invoke(loop_rtcp));
+
+  // Simulate microphone giving audio frame (10 ms). This will trigger transport
+  // to send RTP as handled in loop_rtp above.
+  auto audio_sender = audio_channel_->GetAudioSender();
+  audio_sender->SendAudioData(GetAudioFrame(0));
+  audio_sender->SendAudioData(GetAudioFrame(1));
+
+  // Simulate speaker requesting audio frame (10 ms). This will trigger VoIP
+  // engine to fetch audio samples from RTP packets stored in jitter buffer.
+  AudioFrame audio_frame;
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+  audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+
+  // Force sending RTCP SR report in order to have remote_rtcp field available
+  // in channel statistics. This will trigger transport to send RTCP as handled
+  // in loop_rtcp above.
+  audio_channel_->SendRTCPReportForTesting(kRtcpSr);
+
+  absl::optional<ChannelStatistics> channel_stats =
+      audio_channel_->GetChannelStatistics();
+  EXPECT_TRUE(channel_stats);
+
+  EXPECT_EQ(channel_stats->packets_sent, 1ULL);
+  EXPECT_EQ(channel_stats->bytes_sent, 160ULL);
+
+  EXPECT_EQ(channel_stats->packets_received, 1ULL);
+  EXPECT_EQ(channel_stats->bytes_received, 160ULL);
+  EXPECT_EQ(channel_stats->jitter, 0);
+  EXPECT_EQ(channel_stats->packets_lost, 0);
+  EXPECT_EQ(channel_stats->remote_ssrc.value(), kLocalSsrc);
+
+  EXPECT_TRUE(channel_stats->remote_rtcp.has_value());
+
+  EXPECT_EQ(channel_stats->remote_rtcp->jitter, 0);
+  EXPECT_EQ(channel_stats->remote_rtcp->packets_lost, 0);
+  EXPECT_EQ(channel_stats->remote_rtcp->fraction_lost, 0);
+  EXPECT_GT(channel_stats->remote_rtcp->last_report_received_timestamp_ms, 0);
+  EXPECT_FALSE(channel_stats->remote_rtcp->round_trip_time.has_value());
+}
+
+// Check ChannelStatistics RTT metric after processing RTP and RTCP packets
+// using three audio channels where each represents media endpoint.
+//
+//  1) AC1 <- RTP/RTCP -> AC2
+//  2) AC1 <- RTP/RTCP -> AC3
+//
+// During step 1), AC1 should be able to check RTT from AC2's SSRC.
+// During step 2), AC1 should be able to check RTT from AC3's SSRC.
+TEST_F(AudioChannelTest, RttIsAvailableAfterChangeOfRemoteSsrc) {
+  // Create AC2 and AC3.
+  constexpr uint32_t kAc2Ssrc = 0xdeadbeef;
+  constexpr uint32_t kAc3Ssrc = 0xdeafbeef;
+
+  auto ac_2 = CreateAudioChannel(kAc2Ssrc);
+  auto ac_3 = CreateAudioChannel(kAc3Ssrc);
+
+  auto send_recv_rtp = [&](rtc::scoped_refptr<AudioChannel> rtp_sender,
+                           rtc::scoped_refptr<AudioChannel> rtp_receiver) {
+    // Setup routing logic via transport_.
+    auto route_rtp = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+      rtp_receiver->ReceivedRTPPacket(packet);
+      return true;
+    };
+    ON_CALL(transport_, SendRtp).WillByDefault(route_rtp);
+
+    // This will trigger route_rtp callback via transport_.
+    rtp_sender->GetAudioSender()->SendAudioData(GetAudioFrame(0));
+    rtp_sender->GetAudioSender()->SendAudioData(GetAudioFrame(1));
+
+    // Process received RTP in receiver.
+    AudioFrame audio_frame;
+    audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+    audio_mixer_->Mix(/*number_of_channels=*/1, &audio_frame);
+
+    // Revert to default to avoid using reference in route_rtp lambda.
+    ON_CALL(transport_, SendRtp).WillByDefault(Return(true));
+  };
+
+  auto send_recv_rtcp = [&](rtc::scoped_refptr<AudioChannel> rtcp_sender,
+                            rtc::scoped_refptr<AudioChannel> rtcp_receiver) {
+    // Setup routing logic via transport_.
+    auto route_rtcp = [&](rtc::ArrayView<const uint8_t> packet) {
+      rtcp_receiver->ReceivedRTCPPacket(packet);
+      return true;
+    };
+    ON_CALL(transport_, SendRtcp).WillByDefault(route_rtcp);
+
+    // This will trigger route_rtcp callback via transport_.
+    rtcp_sender->SendRTCPReportForTesting(kRtcpSr);
+
+    // Revert to default to avoid using reference in route_rtcp lambda.
+    ON_CALL(transport_, SendRtcp).WillByDefault(Return(true));
+  };
+
+  // AC1 <-- RTP/RTCP --> AC2
+  send_recv_rtp(audio_channel_, ac_2);
+  send_recv_rtp(ac_2, audio_channel_);
+  send_recv_rtcp(audio_channel_, ac_2);
+  send_recv_rtcp(ac_2, audio_channel_);
+
+  absl::optional<ChannelStatistics> channel_stats =
+      audio_channel_->GetChannelStatistics();
+  ASSERT_TRUE(channel_stats);
+  EXPECT_EQ(channel_stats->remote_ssrc, kAc2Ssrc);
+  ASSERT_TRUE(channel_stats->remote_rtcp);
+  EXPECT_GT(channel_stats->remote_rtcp->round_trip_time, 0.0);
+
+  // AC1 <-- RTP/RTCP --> AC3
+  send_recv_rtp(audio_channel_, ac_3);
+  send_recv_rtp(ac_3, audio_channel_);
+  send_recv_rtcp(audio_channel_, ac_3);
+  send_recv_rtcp(ac_3, audio_channel_);
+
+  channel_stats = audio_channel_->GetChannelStatistics();
+  ASSERT_TRUE(channel_stats);
+  EXPECT_EQ(channel_stats->remote_ssrc, kAc3Ssrc);
+  ASSERT_TRUE(channel_stats->remote_rtcp);
+  EXPECT_GT(channel_stats->remote_rtcp->round_trip_time, 0.0);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/voip/test/audio_egress_unittest.cc b/third_party/libwebrtc/audio/voip/test/audio_egress_unittest.cc
new file mode 100644
index 0000000000..83df26eef1
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/test/audio_egress_unittest.cc
@@ -0,0 +1,326 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/voip/audio_egress.h"
+
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/call/transport.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/audio_mixer/sine_wave_generator.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_transport.h"
+#include "test/run_loop.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Invoke;
+using ::testing::NiceMock;
+using ::testing::Unused;
+
+std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpStack(Clock* clock,
+                                                   Transport* transport,
+                                                   uint32_t remote_ssrc) {
+  RtpRtcpInterface::Configuration rtp_config;
+  rtp_config.clock = clock;
+  rtp_config.audio = true;
+  rtp_config.rtcp_report_interval_ms = 5000;
+  rtp_config.outgoing_transport = transport;
+  rtp_config.local_media_ssrc = remote_ssrc;
+  auto rtp_rtcp = ModuleRtpRtcpImpl2::Create(rtp_config);
+  rtp_rtcp->SetSendingMediaStatus(false);
+  rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
+  return rtp_rtcp;
+}
+
+constexpr int16_t kAudioLevel = 3004;  // Used for sine wave level.
+
+// AudioEgressTest configures audio egress by using Rtp Stack, fake clock,
+// and task queue factory.  Encoder factory is needed to create codec and
+// configure the RTP stack in audio egress.
+class AudioEgressTest : public ::testing::Test {
+ public:
+  static constexpr uint16_t kSeqNum = 12345;
+  static constexpr uint64_t kStartTime = 123456789;
+  static constexpr uint32_t kRemoteSsrc = 0xDEADBEEF;
+  const SdpAudioFormat kPcmuFormat = {"pcmu", 8000, 1};
+
+  AudioEgressTest() : wave_generator_(1000.0, kAudioLevel) {
+    encoder_factory_ = CreateBuiltinAudioEncoderFactory();
+  }
+
+  // Prepare test on audio egress by using PCMu codec with specific
+  // sequence number and its status to be running.
+  void SetUp() override {
+    rtp_rtcp_ =
+        CreateRtpStack(time_controller_.GetClock(), &transport_, kRemoteSsrc);
+    egress_ = std::make_unique<AudioEgress>(
+        rtp_rtcp_.get(), time_controller_.GetClock(),
+        time_controller_.GetTaskQueueFactory());
+    constexpr int kPcmuPayload = 0;
+    egress_->SetEncoder(kPcmuPayload, kPcmuFormat,
+                        encoder_factory_->MakeAudioEncoder(
+                            kPcmuPayload, kPcmuFormat, absl::nullopt));
+    egress_->StartSend();
+    rtp_rtcp_->SetSequenceNumber(kSeqNum);
+    rtp_rtcp_->SetSendingStatus(true);
+  }
+
+  // Make sure we have shut down rtp stack and reset egress for each test.
+  void TearDown() override {
+    egress_->StopSend();
+    rtp_rtcp_->SetSendingStatus(false);
+    egress_.reset();
+    rtp_rtcp_.reset();
+  }
+
+  // Create an audio frame prepared for pcmu encoding. Timestamp is
+  // increased per RTP specification which is the number of samples it contains.
+  // Wave generator writes sine wave which has expected high level set
+  // by kAudioLevel.
+  std::unique_ptr<AudioFrame> GetAudioFrame(int order) {
+    auto frame = std::make_unique<AudioFrame>();
+    frame->sample_rate_hz_ = kPcmuFormat.clockrate_hz;
+    frame->samples_per_channel_ = kPcmuFormat.clockrate_hz / 100;  // 10 ms.
+    frame->num_channels_ = kPcmuFormat.num_channels;
+    frame->timestamp_ = frame->samples_per_channel_ * order;
+    wave_generator_.GenerateNextFrame(frame.get());
+    return frame;
+  }
+
+  GlobalSimulatedTimeController time_controller_{Timestamp::Micros(kStartTime)};
+  NiceMock<MockTransport> transport_;
+  SineWaveGenerator wave_generator_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
+  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
+  std::unique_ptr<AudioEgress> egress_;
+};
+
+TEST_F(AudioEgressTest, SendingStatusAfterStartAndStop) {
+  EXPECT_TRUE(egress_->IsSending());
+  egress_->StopSend();
+  EXPECT_FALSE(egress_->IsSending());
+}
+
+TEST_F(AudioEgressTest, ProcessAudioWithMute) {
+  constexpr int kExpected = 10;
+  rtc::Event event;
+  int rtp_count = 0;
+  RtpPacketReceived rtp;
+  auto rtp_sent = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    rtp.Parse(packet);
+    if (++rtp_count == kExpected) {
+      event.Set();
+    }
+    return true;
+  };
+
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent));
+
+  egress_->SetMute(true);
+
+  // Two 10 ms audio frames will result in rtp packet with ptime 20.
+  for (size_t i = 0; i < kExpected * 2; i++) {
+    egress_->SendAudioData(GetAudioFrame(i));
+    time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  }
+
+  event.Wait(TimeDelta::Seconds(1));
+  EXPECT_EQ(rtp_count, kExpected);
+
+  // we expect on pcmu payload to result in 255 for silenced payload
+  RTPHeader header;
+  rtp.GetHeader(&header);
+  size_t packet_length = rtp.size();
+  size_t payload_length = packet_length - header.headerLength;
+  size_t payload_data_length = payload_length - header.paddingLength;
+  const uint8_t* payload = rtp.data() + header.headerLength;
+  for (size_t i = 0; i < payload_data_length; ++i) {
+    EXPECT_EQ(*payload++, 255);
+  }
+}
+
+TEST_F(AudioEgressTest, ProcessAudioWithSineWave) {
+  constexpr int kExpected = 10;
+  rtc::Event event;
+  int rtp_count = 0;
+  RtpPacketReceived rtp;
+  auto rtp_sent = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    rtp.Parse(packet);
+    if (++rtp_count == kExpected) {
+      event.Set();
+    }
+    return true;
+  };
+
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent));
+
+  // Two 10 ms audio frames will result in rtp packet with ptime 20.
+  for (size_t i = 0; i < kExpected * 2; i++) {
+    egress_->SendAudioData(GetAudioFrame(i));
+    time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  }
+
+  event.Wait(TimeDelta::Seconds(1));
+  EXPECT_EQ(rtp_count, kExpected);
+
+  // we expect on pcmu to result in < 255 for payload with sine wave
+  RTPHeader header;
+  rtp.GetHeader(&header);
+  size_t packet_length = rtp.size();
+  size_t payload_length = packet_length - header.headerLength;
+  size_t payload_data_length = payload_length - header.paddingLength;
+  const uint8_t* payload = rtp.data() + header.headerLength;
+  for (size_t i = 0; i < payload_data_length; ++i) {
+    EXPECT_NE(*payload++, 255);
+  }
+}
+
+TEST_F(AudioEgressTest, SkipAudioEncodingAfterStopSend) {
+  constexpr int kExpected = 10;
+  rtc::Event event;
+  int rtp_count = 0;
+  auto rtp_sent = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    if (++rtp_count == kExpected) {
+      event.Set();
+    }
+    return true;
+  };
+
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent));
+
+  // Two 10 ms audio frames will result in rtp packet with ptime 20.
+  for (size_t i = 0; i < kExpected * 2; i++) {
+    egress_->SendAudioData(GetAudioFrame(i));
+    time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  }
+
+  event.Wait(TimeDelta::Seconds(1));
+  EXPECT_EQ(rtp_count, kExpected);
+
+  // Now stop send and yet feed more data.
+  egress_->StopSend();
+
+  // It should be safe to exit the test case while encoder_queue_ has
+  // outstanding data to process. We are making sure that this doesn't
+  // result in crashes or sanitizer errors due to remaining data.
+  for (size_t i = 0; i < kExpected * 2; i++) {
+    egress_->SendAudioData(GetAudioFrame(i));
+    time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  }
+}
+
+TEST_F(AudioEgressTest, ChangeEncoderFromPcmuToOpus) {
+  absl::optional<SdpAudioFormat> pcmu = egress_->GetEncoderFormat();
+  EXPECT_TRUE(pcmu);
+  EXPECT_EQ(pcmu->clockrate_hz, kPcmuFormat.clockrate_hz);
+  EXPECT_EQ(pcmu->num_channels, kPcmuFormat.num_channels);
+
+  constexpr int kOpusPayload = 120;
+  const SdpAudioFormat kOpusFormat = {"opus", 48000, 2};
+
+  egress_->SetEncoder(kOpusPayload, kOpusFormat,
+                      encoder_factory_->MakeAudioEncoder(
+                          kOpusPayload, kOpusFormat, absl::nullopt));
+
+  absl::optional<SdpAudioFormat> opus = egress_->GetEncoderFormat();
+  EXPECT_TRUE(opus);
+  EXPECT_EQ(opus->clockrate_hz, kOpusFormat.clockrate_hz);
+  EXPECT_EQ(opus->num_channels, kOpusFormat.num_channels);
+}
+
+TEST_F(AudioEgressTest, SendDTMF) {
+  constexpr int kExpected = 7;
+  constexpr int kPayloadType = 100;
+  constexpr int kDurationMs = 100;
+  constexpr int kSampleRate = 8000;
+  constexpr int kEvent = 3;
+
+  egress_->RegisterTelephoneEventType(kPayloadType, kSampleRate);
+  // 100 ms duration will produce total 7 DTMF
+  // 1 @ 20 ms, 2 @ 40 ms, 3 @ 60 ms, 4 @ 80 ms
+  // 5, 6, 7 @ 100 ms (last one sends 3 dtmf)
+  egress_->SendTelephoneEvent(kEvent, kDurationMs);
+
+  rtc::Event event;
+  int dtmf_count = 0;
+  auto is_dtmf = [&](RtpPacketReceived& rtp) {
+    return (rtp.PayloadType() == kPayloadType &&
+            rtp.SequenceNumber() == kSeqNum + dtmf_count &&
+            rtp.padding_size() == 0 && rtp.Marker() == (dtmf_count == 0) &&
+            rtp.Ssrc() == kRemoteSsrc);
+  };
+
+  // It's possible that we may have actual audio RTP packets along with
+  // DTMF packtets.  We are only interested in the exact number of DTMF
+  // packets rtp stack is emitting.
+  auto rtp_sent = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    RtpPacketReceived rtp;
+    rtp.Parse(packet);
+    if (is_dtmf(rtp) && ++dtmf_count == kExpected) {
+      event.Set();
+    }
+    return true;
+  };
+
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent));
+
+  // Two 10 ms audio frames will result in rtp packet with ptime 20.
+  for (size_t i = 0; i < kExpected * 2; i++) {
+    egress_->SendAudioData(GetAudioFrame(i));
+    time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  }
+
+  event.Wait(TimeDelta::Seconds(1));
+  EXPECT_EQ(dtmf_count, kExpected);
+}
+
+TEST_F(AudioEgressTest, TestAudioInputLevelAndEnergyDuration) {
+  // Per audio_level's kUpdateFrequency, we need more than 10 audio samples to
+  // get audio level from input source.
+  constexpr int kExpected = 6;
+  rtc::Event event;
+  int rtp_count = 0;
+  auto rtp_sent = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    if (++rtp_count == kExpected) {
+      event.Set();
+    }
+    return true;
+  };
+
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent));
+
+  // Two 10 ms audio frames will result in rtp packet with ptime 20.
+  for (size_t i = 0; i < kExpected * 2; i++) {
+    egress_->SendAudioData(GetAudioFrame(i));
+    time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  }
+
+  event.Wait(/*give_up_after=*/TimeDelta::Seconds(1));
+  EXPECT_EQ(rtp_count, kExpected);
+
+  constexpr double kExpectedEnergy = 0.00016809565587789564;
+  constexpr double kExpectedDuration = 0.11999999999999998;
+
+  EXPECT_EQ(egress_->GetInputAudioLevel(), kAudioLevel);
+  EXPECT_DOUBLE_EQ(egress_->GetInputTotalEnergy(), kExpectedEnergy);
+  EXPECT_DOUBLE_EQ(egress_->GetInputTotalDuration(), kExpectedDuration);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/voip/test/audio_ingress_unittest.cc b/third_party/libwebrtc/audio/voip/test/audio_ingress_unittest.cc
new file mode 100644
index 0000000000..c7736b247a
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/test/audio_ingress_unittest.cc
@@ -0,0 +1,240 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/voip/audio_ingress.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/call/transport.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/units/time_delta.h"
+#include "audio/voip/audio_egress.h"
+#include "modules/audio_mixer/sine_wave_generator.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_transport.h"
+#include "test/run_loop.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Invoke;
+using ::testing::NiceMock;
+using ::testing::Unused;
+
+constexpr int16_t kAudioLevel = 3004;  // Used for sine wave level.
+
+class AudioIngressTest : public ::testing::Test {
+ public:
+  const SdpAudioFormat kPcmuFormat = {"pcmu", 8000, 1};
+
+  AudioIngressTest() : wave_generator_(1000.0, kAudioLevel) {
+    receive_statistics_ =
+        ReceiveStatistics::Create(time_controller_.GetClock());
+
+    RtpRtcpInterface::Configuration rtp_config;
+    rtp_config.clock = time_controller_.GetClock();
+    rtp_config.audio = true;
+    rtp_config.receive_statistics = receive_statistics_.get();
+    rtp_config.rtcp_report_interval_ms = 5000;
+    rtp_config.outgoing_transport = &transport_;
+    rtp_config.local_media_ssrc = 0xdeadc0de;
+    rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
+
+    rtp_rtcp_->SetSendingMediaStatus(false);
+    rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
+
+    encoder_factory_ = CreateBuiltinAudioEncoderFactory();
+    decoder_factory_ = CreateBuiltinAudioDecoderFactory();
+  }
+
+  void SetUp() override {
+    constexpr int kPcmuPayload = 0;
+    ingress_ = std::make_unique<AudioIngress>(
+        rtp_rtcp_.get(), time_controller_.GetClock(), receive_statistics_.get(),
+        decoder_factory_);
+    ingress_->SetReceiveCodecs({{kPcmuPayload, kPcmuFormat}});
+
+    egress_ = std::make_unique<AudioEgress>(
+        rtp_rtcp_.get(), time_controller_.GetClock(),
+        time_controller_.GetTaskQueueFactory());
+    egress_->SetEncoder(kPcmuPayload, kPcmuFormat,
+                        encoder_factory_->MakeAudioEncoder(
+                            kPcmuPayload, kPcmuFormat, absl::nullopt));
+    egress_->StartSend();
+    ingress_->StartPlay();
+    rtp_rtcp_->SetSendingStatus(true);
+  }
+
+  void TearDown() override {
+    rtp_rtcp_->SetSendingStatus(false);
+    ingress_->StopPlay();
+    egress_->StopSend();
+    egress_.reset();
+    ingress_.reset();
+  }
+
+  std::unique_ptr<AudioFrame> GetAudioFrame(int order) {
+    auto frame = std::make_unique<AudioFrame>();
+    frame->sample_rate_hz_ = kPcmuFormat.clockrate_hz;
+    frame->samples_per_channel_ = kPcmuFormat.clockrate_hz / 100;  // 10 ms.
+    frame->num_channels_ = kPcmuFormat.num_channels;
+    frame->timestamp_ = frame->samples_per_channel_ * order;
+    wave_generator_.GenerateNextFrame(frame.get());
+    return frame;
+  }
+
+  GlobalSimulatedTimeController time_controller_{Timestamp::Micros(123456789)};
+  SineWaveGenerator wave_generator_;
+  NiceMock<MockTransport> transport_;
+  std::unique_ptr<ReceiveStatistics> receive_statistics_;
+  std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
+  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  std::unique_ptr<AudioIngress> ingress_;
+  std::unique_ptr<AudioEgress> egress_;
+};
+
+TEST_F(AudioIngressTest, PlayingAfterStartAndStop) {
+  EXPECT_EQ(ingress_->IsPlaying(), true);
+  ingress_->StopPlay();
+  EXPECT_EQ(ingress_->IsPlaying(), false);
+}
+
+TEST_F(AudioIngressTest, GetAudioFrameAfterRtpReceived) {
+  rtc::Event event;
+  auto handle_rtp = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    ingress_->ReceivedRTPPacket(packet);
+    event.Set();
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp));
+  egress_->SendAudioData(GetAudioFrame(0));
+  egress_->SendAudioData(GetAudioFrame(1));
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+  ASSERT_TRUE(event.Wait(TimeDelta::Seconds(1)));
+
+  AudioFrame audio_frame;
+  EXPECT_EQ(
+      ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame),
+      AudioMixer::Source::AudioFrameInfo::kNormal);
+  EXPECT_FALSE(audio_frame.muted());
+  EXPECT_EQ(audio_frame.num_channels_, 1u);
+  EXPECT_EQ(audio_frame.samples_per_channel_,
+            static_cast<size_t>(kPcmuFormat.clockrate_hz / 100));
+  EXPECT_EQ(audio_frame.sample_rate_hz_, kPcmuFormat.clockrate_hz);
+  EXPECT_NE(audio_frame.timestamp_, 0u);
+  EXPECT_EQ(audio_frame.elapsed_time_ms_, 0);
+}
+
+TEST_F(AudioIngressTest, TestSpeechOutputLevelAndEnergyDuration) {
+  // Per audio_level's kUpdateFrequency, we need more than 10 audio samples to
+  // get audio level from output source.
+  constexpr int kNumRtp = 6;
+  int rtp_count = 0;
+  rtc::Event event;
+  auto handle_rtp = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    ingress_->ReceivedRTPPacket(packet);
+    if (++rtp_count == kNumRtp) {
+      event.Set();
+    }
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp));
+  for (int i = 0; i < kNumRtp * 2; i++) {
+    egress_->SendAudioData(GetAudioFrame(i));
+    time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  }
+  event.Wait(/*give_up_after=*/TimeDelta::Seconds(1));
+
+  for (int i = 0; i < kNumRtp * 2; ++i) {
+    AudioFrame audio_frame;
+    EXPECT_EQ(
+        ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame),
+        AudioMixer::Source::AudioFrameInfo::kNormal);
+  }
+  EXPECT_EQ(ingress_->GetOutputAudioLevel(), kAudioLevel);
+
+  constexpr double kExpectedEnergy = 0.00016809565587789564;
+  constexpr double kExpectedDuration = 0.11999999999999998;
+
+  EXPECT_DOUBLE_EQ(ingress_->GetOutputTotalEnergy(), kExpectedEnergy);
+  EXPECT_DOUBLE_EQ(ingress_->GetOutputTotalDuration(), kExpectedDuration);
+}
+
+TEST_F(AudioIngressTest, PreferredSampleRate) {
+  rtc::Event event;
+  auto handle_rtp = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    ingress_->ReceivedRTPPacket(packet);
+    event.Set();
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp));
+  egress_->SendAudioData(GetAudioFrame(0));
+  egress_->SendAudioData(GetAudioFrame(1));
+  time_controller_.AdvanceTime(TimeDelta::Zero());
+  ASSERT_TRUE(event.Wait(TimeDelta::Seconds(1)));
+
+  AudioFrame audio_frame;
+  EXPECT_EQ(
+      ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame),
+      AudioMixer::Source::AudioFrameInfo::kNormal);
+  EXPECT_EQ(ingress_->PreferredSampleRate(), kPcmuFormat.clockrate_hz);
+}
+
+// This test highlights the case where caller invokes StopPlay() which then
+// AudioIngress should play silence frame afterwards.
+TEST_F(AudioIngressTest, GetMutedAudioFrameAfterRtpReceivedAndStopPlay) {
+  // StopPlay before we start sending RTP packet with sine wave.
+  ingress_->StopPlay();
+
+  // Send 6 RTP packets to generate more than 100 ms audio sample to get
+  // valid speech level.
+  constexpr int kNumRtp = 6;
+  int rtp_count = 0;
+  rtc::Event event;
+  auto handle_rtp = [&](rtc::ArrayView<const uint8_t> packet, Unused) {
+    ingress_->ReceivedRTPPacket(packet);
+    if (++rtp_count == kNumRtp) {
+      event.Set();
+    }
+    return true;
+  };
+  EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp));
+  for (int i = 0; i < kNumRtp * 2; i++) {
+    egress_->SendAudioData(GetAudioFrame(i));
+    time_controller_.AdvanceTime(TimeDelta::Millis(10));
+  }
+  event.Wait(/*give_up_after=*/TimeDelta::Seconds(1));
+
+  for (int i = 0; i < kNumRtp * 2; ++i) {
+    AudioFrame audio_frame;
+    EXPECT_EQ(
+        ingress_->GetAudioFrameWithInfo(kPcmuFormat.clockrate_hz, &audio_frame),
+        AudioMixer::Source::AudioFrameInfo::kMuted);
+    const int16_t* audio_data = audio_frame.data();
+    size_t length =
+        audio_frame.samples_per_channel_ * audio_frame.num_channels_;
+    for (size_t j = 0; j < length; ++j) {
+      EXPECT_EQ(audio_data[j], 0);
+    }
+  }
+
+  // Now we should still see valid speech output level as StopPlay won't affect
+  // the measurement.
+  EXPECT_EQ(ingress_->GetOutputAudioLevel(), kAudioLevel);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/voip/test/mock_task_queue.h b/third_party/libwebrtc/audio/voip/test/mock_task_queue.h
new file mode 100644
index 0000000000..547b0d3f75
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/test/mock_task_queue.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_VOIP_TEST_MOCK_TASK_QUEUE_H_
+#define AUDIO_VOIP_TEST_MOCK_TASK_QUEUE_H_
+
+#include <memory>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "api/task_queue/test/mock_task_queue_base.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+// MockTaskQueue enables immediate task run from global TaskQueueBase.
+// It's necessary for some tests depending on TaskQueueBase internally.
+class MockTaskQueue : public MockTaskQueueBase {
+ public:
+  MockTaskQueue() : current_(this) {}
+
+  // Delete is deliberately defined as no-op as MockTaskQueue is expected to
+  // hold onto current global TaskQueueBase throughout the testing.
+  void Delete() override {}
+
+ private:
+  CurrentTaskQueueSetter current_;
+};
+
+class MockTaskQueueFactory : public TaskQueueFactory {
+ public:
+  explicit MockTaskQueueFactory(MockTaskQueue* task_queue)
+      : task_queue_(task_queue) {}
+
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
+      absl::string_view name,
+      Priority priority) const override {
+    // Default MockTaskQueue::Delete is no-op, therefore it's safe to pass the
+    // raw pointer.
+    return std::unique_ptr<TaskQueueBase, TaskQueueDeleter>(task_queue_);
+  }
+
+ private:
+  MockTaskQueue* task_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_TEST_MOCK_TASK_QUEUE_H_
diff --git a/third_party/libwebrtc/audio/voip/test/voip_core_unittest.cc b/third_party/libwebrtc/audio/voip/test/voip_core_unittest.cc
new file mode 100644
index 0000000000..b432506b12
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/test/voip_core_unittest.cc
@@ -0,0 +1,193 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/voip/voip_core.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/audio_processing/include/mock_audio_processing.h"
+#include "test/gtest.h"
+#include "test/mock_transport.h"
+#include "test/run_loop.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::NiceMock;
+using ::testing::Return;
+
+constexpr int kPcmuPayload = 0;
+constexpr int kPcmuSampleRateHz = 8000;
+constexpr int kDtmfEventDurationMs = 1000;
+constexpr DtmfEvent kDtmfEventCode = DtmfEvent::kDigitZero;
+
+class VoipCoreTest : public ::testing::Test {
+ public:
+  const SdpAudioFormat kPcmuFormat = {"pcmu", 8000, 1};
+
+  VoipCoreTest() { audio_device_ = test::MockAudioDeviceModule::CreateNice(); }
+
+  void SetUp() override {
+    auto encoder_factory = CreateBuiltinAudioEncoderFactory();
+    auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+    rtc::scoped_refptr<AudioProcessing> audio_processing =
+        rtc::make_ref_counted<NiceMock<test::MockAudioProcessing>>();
+
+    voip_core_ = std::make_unique<VoipCore>(
+        std::move(encoder_factory), std::move(decoder_factory),
+        CreateDefaultTaskQueueFactory(), audio_device_,
+        std::move(audio_processing));
+  }
+
+  test::RunLoop run_loop_;
+  std::unique_ptr<VoipCore> voip_core_;
+  NiceMock<MockTransport> transport_;
+  rtc::scoped_refptr<test::MockAudioDeviceModule> audio_device_;
+};
+
+// Validate expected API calls that involves with VoipCore. Some verification is
+// involved with checking mock audio device.
+TEST_F(VoipCoreTest, BasicVoipCoreOperation) {
+  // Program mock as non-operational and ready to start.
+  EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(false));
+  EXPECT_CALL(*audio_device_, Playing()).WillOnce(Return(false));
+  EXPECT_CALL(*audio_device_, InitRecording()).WillOnce(Return(0));
+  EXPECT_CALL(*audio_device_, InitPlayout()).WillOnce(Return(0));
+  EXPECT_CALL(*audio_device_, StartRecording()).WillOnce(Return(0));
+  EXPECT_CALL(*audio_device_, StartPlayout()).WillOnce(Return(0));
+
+  auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
+
+  EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat),
+            VoipResult::kOk);
+  EXPECT_EQ(
+      voip_core_->SetReceiveCodecs(channel, {{kPcmuPayload, kPcmuFormat}}),
+      VoipResult::kOk);
+
+  EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kOk);
+  EXPECT_EQ(voip_core_->StartPlayout(channel), VoipResult::kOk);
+
+  EXPECT_EQ(voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload,
+                                                   kPcmuSampleRateHz),
+            VoipResult::kOk);
+
+  EXPECT_EQ(
+      voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs),
+      VoipResult::kOk);
+
+  // Program mock as operational that is ready to be stopped.
+  EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(true));
+  EXPECT_CALL(*audio_device_, Playing()).WillOnce(Return(true));
+  EXPECT_CALL(*audio_device_, StopRecording()).WillOnce(Return(0));
+  EXPECT_CALL(*audio_device_, StopPlayout()).WillOnce(Return(0));
+
+  EXPECT_EQ(voip_core_->StopSend(channel), VoipResult::kOk);
+  EXPECT_EQ(voip_core_->StopPlayout(channel), VoipResult::kOk);
+  EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
+}
+
+TEST_F(VoipCoreTest, ExpectFailToUseReleasedChannelId) {
+  auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
+
+  // Release right after creation.
+  EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
+
+  // Now use released channel.
+
+  EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat),
+            VoipResult::kInvalidArgument);
+  EXPECT_EQ(
+      voip_core_->SetReceiveCodecs(channel, {{kPcmuPayload, kPcmuFormat}}),
+      VoipResult::kInvalidArgument);
+  EXPECT_EQ(voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload,
+                                                   kPcmuSampleRateHz),
+            VoipResult::kInvalidArgument);
+  EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kInvalidArgument);
+  EXPECT_EQ(voip_core_->StartPlayout(channel), VoipResult::kInvalidArgument);
+  EXPECT_EQ(
+      voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs),
+      VoipResult::kInvalidArgument);
+}
+
+TEST_F(VoipCoreTest, SendDtmfEventWithoutRegistering) {
+  // Program mock as non-operational and ready to start send.
+  EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(false));
+  EXPECT_CALL(*audio_device_, InitRecording()).WillOnce(Return(0));
+  EXPECT_CALL(*audio_device_, StartRecording()).WillOnce(Return(0));
+
+  auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
+
+  EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat),
+            VoipResult::kOk);
+
+  EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kOk);
+  // Send Dtmf event without registering beforehand, thus payload
+  // type is not set and kFailedPrecondition is expected.
+  EXPECT_EQ(
+      voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs),
+      VoipResult::kFailedPrecondition);
+
+  // Program mock as sending and is ready to be stopped.
+  EXPECT_CALL(*audio_device_, Recording()).WillOnce(Return(true));
+  EXPECT_CALL(*audio_device_, StopRecording()).WillOnce(Return(0));
+
+  EXPECT_EQ(voip_core_->StopSend(channel), VoipResult::kOk);
+  EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
+}
+
+TEST_F(VoipCoreTest, SendDtmfEventWithoutStartSend) {
+  auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
+
+  EXPECT_EQ(voip_core_->RegisterTelephoneEventType(channel, kPcmuPayload,
+                                                   kPcmuSampleRateHz),
+            VoipResult::kOk);
+
+  // Send Dtmf event without calling StartSend beforehand, thus
+  // Dtmf events cannot be sent and kFailedPrecondition is expected.
+  EXPECT_EQ(
+      voip_core_->SendDtmfEvent(channel, kDtmfEventCode, kDtmfEventDurationMs),
+      VoipResult::kFailedPrecondition);
+
+  EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
+}
+
+TEST_F(VoipCoreTest, StartSendAndPlayoutWithoutSettingCodec) {
+  auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
+
+  // Call StartSend and StartPlayout without setting send/receive
+  // codec. Code should see that codecs aren't set and return false.
+  EXPECT_EQ(voip_core_->StartSend(channel), VoipResult::kFailedPrecondition);
+  EXPECT_EQ(voip_core_->StartPlayout(channel), VoipResult::kFailedPrecondition);
+
+  EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
+}
+
+TEST_F(VoipCoreTest, StopSendAndPlayoutWithoutStarting) {
+  auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de);
+
+  EXPECT_EQ(voip_core_->SetSendCodec(channel, kPcmuPayload, kPcmuFormat),
+            VoipResult::kOk);
+  EXPECT_EQ(
+      voip_core_->SetReceiveCodecs(channel, {{kPcmuPayload, kPcmuFormat}}),
+      VoipResult::kOk);
+
+  // Call StopSend and StopPlayout without starting them in
+  // the first place. Should see that it is already in the
+  // stopped state and return true.
+  EXPECT_EQ(voip_core_->StopSend(channel), VoipResult::kOk);
+  EXPECT_EQ(voip_core_->StopPlayout(channel), VoipResult::kOk);
+
+  EXPECT_EQ(voip_core_->ReleaseChannel(channel), VoipResult::kOk);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/voip/voip_core.cc b/third_party/libwebrtc/audio/voip/voip_core.cc
new file mode 100644
index 0000000000..8df1c594aa
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/voip_core.cc
@@ -0,0 +1,500 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/voip/voip_core.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "api/audio_codecs/audio_format.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+// For Windows, use specific enum type to initialize default audio device as
+// defined in AudioDeviceModule::WindowsDeviceType.
+#if defined(WEBRTC_WIN)
+constexpr AudioDeviceModule::WindowsDeviceType kAudioDeviceId =
+    AudioDeviceModule::WindowsDeviceType::kDefaultCommunicationDevice;
+#else
+constexpr uint16_t kAudioDeviceId = 0;
+#endif  // defined(WEBRTC_WIN)
+
+// Maximum value range limit on ChannelId. This can be increased without any
+// side effect and only set at this moderate value for better readability for
+// logging.
+static constexpr int kMaxChannelId = 100000;
+
+}  // namespace
+
+VoipCore::VoipCore(rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
+                   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+                   std::unique_ptr<TaskQueueFactory> task_queue_factory,
+                   rtc::scoped_refptr<AudioDeviceModule> audio_device_module,
+                   rtc::scoped_refptr<AudioProcessing> audio_processing) {
+  encoder_factory_ = std::move(encoder_factory);
+  decoder_factory_ = std::move(decoder_factory);
+  task_queue_factory_ = std::move(task_queue_factory);
+  audio_device_module_ = std::move(audio_device_module);
+  audio_processing_ = std::move(audio_processing);
+  audio_mixer_ = AudioMixerImpl::Create();
+
+  // AudioTransportImpl depends on audio mixer and audio processing instances.
+  audio_transport_ = std::make_unique<AudioTransportImpl>(
+      audio_mixer_.get(), audio_processing_.get(), nullptr);
+}
+
+bool VoipCore::InitializeIfNeeded() {
+  // `audio_device_module_` internally owns a lock and the whole logic here
+  // needs to be executed atomically once using another lock in VoipCore.
+  // Further changes in this method will need to make sure that no deadlock is
+  // introduced in the future.
+  MutexLock lock(&lock_);
+
+  if (initialized_) {
+    return true;
+  }
+
+  // Initialize ADM.
+  if (audio_device_module_->Init() != 0) {
+    RTC_LOG(LS_ERROR) << "Failed to initialize the ADM.";
+    return false;
+  }
+
+  // Note that failures on initializing default recording/speaker devices are
+  // not considered to be fatal here. In certain case, caller may not care about
+  // recording device functioning (e.g webinar where only speaker is available).
+  // It's also possible that there are other audio devices available that may
+  // work.
+
+  // Initialize default speaker device.
+  if (audio_device_module_->SetPlayoutDevice(kAudioDeviceId) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to set playout device.";
+  }
+  if (audio_device_module_->InitSpeaker() != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to access speaker.";
+  }
+
+  // Initialize default recording device.
+  if (audio_device_module_->SetRecordingDevice(kAudioDeviceId) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to set recording device.";
+  }
+  if (audio_device_module_->InitMicrophone() != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to access microphone.";
+  }
+
+  // Set number of channels on speaker device.
+  bool available = false;
+  if (audio_device_module_->StereoPlayoutIsAvailable(&available) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to query stereo playout.";
+  }
+  if (audio_device_module_->SetStereoPlayout(available) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to set mono/stereo playout mode.";
+  }
+
+  // Set number of channels on recording device.
+  available = false;
+  if (audio_device_module_->StereoRecordingIsAvailable(&available) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to query stereo recording.";
+  }
+  if (audio_device_module_->SetStereoRecording(available) != 0) {
+    RTC_LOG(LS_WARNING) << "Unable to set stereo recording mode.";
+  }
+
+  if (audio_device_module_->RegisterAudioCallback(audio_transport_.get()) !=
+      0) {
+    RTC_LOG(LS_WARNING) << "Unable to register audio callback.";
+  }
+
+  initialized_ = true;
+
+  return true;
+}
+
+ChannelId VoipCore::CreateChannel(Transport* transport,
+                                  absl::optional<uint32_t> local_ssrc) {
+  ChannelId channel_id;
+
+  // Set local ssrc to random if not set by caller.
+  if (!local_ssrc) {
+    Random random(rtc::TimeMicros());
+    local_ssrc = random.Rand<uint32_t>();
+  }
+
+  rtc::scoped_refptr<AudioChannel> channel =
+      rtc::make_ref_counted<AudioChannel>(transport, local_ssrc.value(),
+                                          task_queue_factory_.get(),
+                                          audio_mixer_.get(), decoder_factory_);
+
+  {
+    MutexLock lock(&lock_);
+
+    channel_id = static_cast<ChannelId>(next_channel_id_);
+    channels_[channel_id] = channel;
+    next_channel_id_++;
+    if (next_channel_id_ >= kMaxChannelId) {
+      next_channel_id_ = 0;
+    }
+  }
+
+  // Set ChannelId in audio channel for logging/debugging purpose.
+  channel->SetId(channel_id);
+
+  return channel_id;
+}
+
+VoipResult VoipCore::ReleaseChannel(ChannelId channel_id) {
+  // Destroy channel outside of the lock.
+  rtc::scoped_refptr<AudioChannel> channel;
+
+  bool no_channels_after_release = false;
+
+  {
+    MutexLock lock(&lock_);
+
+    auto iter = channels_.find(channel_id);
+    if (iter != channels_.end()) {
+      channel = std::move(iter->second);
+      channels_.erase(iter);
+    }
+
+    no_channels_after_release = channels_.empty();
+  }
+
+  VoipResult status_code = VoipResult::kOk;
+  if (!channel) {
+    RTC_LOG(LS_WARNING) << "Channel " << channel_id << " not found";
+    status_code = VoipResult::kInvalidArgument;
+  }
+
+  if (no_channels_after_release) {
+    // TODO(bugs.webrtc.org/11581): unclear if we still need to clear `channel`
+    // here.
+    channel = nullptr;
+
+    // Make sure to stop playout on ADM if it is playing.
+    if (audio_device_module_->Playing()) {
+      if (audio_device_module_->StopPlayout() != 0) {
+        RTC_LOG(LS_WARNING) << "StopPlayout failed";
+        status_code = VoipResult::kInternal;
+      }
+    }
+  }
+
+  return status_code;
+}
+
+rtc::scoped_refptr<AudioChannel> VoipCore::GetChannel(ChannelId channel_id) {
+  rtc::scoped_refptr<AudioChannel> channel;
+  {
+    MutexLock lock(&lock_);
+    auto iter = channels_.find(channel_id);
+    if (iter != channels_.end()) {
+      channel = iter->second;
+    }
+  }
+  if (!channel) {
+    RTC_LOG(LS_ERROR) << "Channel " << channel_id << " not found";
+  }
+  return channel;
+}
+
+bool VoipCore::UpdateAudioTransportWithSenders() {
+  std::vector<AudioSender*> audio_senders;
+
+  // Gather a list of audio channel that are currently sending along with
+  // highest sampling rate and channel numbers to configure into audio
+  // transport.
+  int max_sampling_rate = 8000;
+  size_t max_num_channels = 1;
+  {
+    MutexLock lock(&lock_);
+    // Reserve to prevent run time vector re-allocation.
+    audio_senders.reserve(channels_.size());
+    for (auto kv : channels_) {
+      rtc::scoped_refptr<AudioChannel>& channel = kv.second;
+      if (channel->IsSendingMedia()) {
+        auto encoder_format = channel->GetEncoderFormat();
+        if (!encoder_format) {
+          RTC_LOG(LS_ERROR)
+              << "channel " << channel->GetId() << " encoder is not set";
+          continue;
+        }
+        audio_senders.push_back(channel->GetAudioSender());
+        max_sampling_rate =
+            std::max(max_sampling_rate, encoder_format->clockrate_hz);
+        max_num_channels =
+            std::max(max_num_channels, encoder_format->num_channels);
+      }
+    }
+  }
+
+  audio_transport_->UpdateAudioSenders(audio_senders, max_sampling_rate,
+                                       max_num_channels);
+
+  // Depending on availability of senders, turn on or off ADM recording.
+  if (!audio_senders.empty()) {
+    // Initialize audio device module and default device if needed.
+    if (!InitializeIfNeeded()) {
+      return false;
+    }
+
+    if (!audio_device_module_->Recording()) {
+      if (audio_device_module_->InitRecording() != 0) {
+        RTC_LOG(LS_ERROR) << "InitRecording failed";
+        return false;
+      }
+      if (audio_device_module_->StartRecording() != 0) {
+        RTC_LOG(LS_ERROR) << "StartRecording failed";
+        return false;
+      }
+    }
+  } else {
+    if (audio_device_module_->Recording() &&
+        audio_device_module_->StopRecording() != 0) {
+      RTC_LOG(LS_ERROR) << "StopRecording failed";
+      return false;
+    }
+  }
+  return true;
+}
+
+VoipResult VoipCore::StartSend(ChannelId channel_id) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  if (!channel->StartSend()) {
+    return VoipResult::kFailedPrecondition;
+  }
+
+  return UpdateAudioTransportWithSenders() ? VoipResult::kOk
+                                           : VoipResult::kInternal;
+}
+
+VoipResult VoipCore::StopSend(ChannelId channel_id) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->StopSend();
+
+  return UpdateAudioTransportWithSenders() ? VoipResult::kOk
+                                           : VoipResult::kInternal;
+}
+
+VoipResult VoipCore::StartPlayout(ChannelId channel_id) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  if (channel->IsPlaying()) {
+    return VoipResult::kOk;
+  }
+
+  if (!channel->StartPlay()) {
+    return VoipResult::kFailedPrecondition;
+  }
+
+  // Initialize audio device module and default device if needed.
+  if (!InitializeIfNeeded()) {
+    return VoipResult::kInternal;
+  }
+
+  if (!audio_device_module_->Playing()) {
+    if (audio_device_module_->InitPlayout() != 0) {
+      RTC_LOG(LS_ERROR) << "InitPlayout failed";
+      return VoipResult::kInternal;
+    }
+    if (audio_device_module_->StartPlayout() != 0) {
+      RTC_LOG(LS_ERROR) << "StartPlayout failed";
+      return VoipResult::kInternal;
+    }
+  }
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::StopPlayout(ChannelId channel_id) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->StopPlay();
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::ReceivedRTPPacket(
+    ChannelId channel_id,
+    rtc::ArrayView<const uint8_t> rtp_packet) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->ReceivedRTPPacket(rtp_packet);
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::ReceivedRTCPPacket(
+    ChannelId channel_id,
+    rtc::ArrayView<const uint8_t> rtcp_packet) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->ReceivedRTCPPacket(rtcp_packet);
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::SetSendCodec(ChannelId channel_id,
+                                  int payload_type,
+                                  const SdpAudioFormat& encoder_format) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  auto encoder = encoder_factory_->MakeAudioEncoder(
+      payload_type, encoder_format, absl::nullopt);
+  channel->SetEncoder(payload_type, encoder_format, std::move(encoder));
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::SetReceiveCodecs(
+    ChannelId channel_id,
+    const std::map<int, SdpAudioFormat>& decoder_specs) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->SetReceiveCodecs(decoder_specs);
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::RegisterTelephoneEventType(ChannelId channel_id,
+                                                int rtp_payload_type,
+                                                int sample_rate_hz) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->RegisterTelephoneEventType(rtp_payload_type, sample_rate_hz);
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::SendDtmfEvent(ChannelId channel_id,
+                                   DtmfEvent dtmf_event,
+                                   int duration_ms) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  return (channel->SendTelephoneEvent(static_cast<int>(dtmf_event), duration_ms)
+              ? VoipResult::kOk
+              : VoipResult::kFailedPrecondition);
+}
+
+VoipResult VoipCore::GetIngressStatistics(ChannelId channel_id,
+                                          IngressStatistics& ingress_stats) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  ingress_stats = channel->GetIngressStatistics();
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::GetChannelStatistics(ChannelId channel_id,
+                                          ChannelStatistics& channel_stats) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel_stats = channel->GetChannelStatistics();
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::SetInputMuted(ChannelId channel_id, bool enable) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  channel->SetMute(enable);
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::GetInputVolumeInfo(ChannelId channel_id,
+                                        VolumeInfo& input_volume) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  input_volume.audio_level = channel->GetInputAudioLevel();
+  input_volume.total_energy = channel->GetInputTotalEnergy();
+  input_volume.total_duration = channel->GetInputTotalDuration();
+
+  return VoipResult::kOk;
+}
+
+VoipResult VoipCore::GetOutputVolumeInfo(ChannelId channel_id,
+                                         VolumeInfo& output_volume) {
+  rtc::scoped_refptr<AudioChannel> channel = GetChannel(channel_id);
+
+  if (!channel) {
+    return VoipResult::kInvalidArgument;
+  }
+
+  output_volume.audio_level = channel->GetOutputAudioLevel();
+  output_volume.total_energy = channel->GetOutputTotalEnergy();
+  output_volume.total_duration = channel->GetOutputTotalDuration();
+
+  return VoipResult::kOk;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/voip/voip_core.h b/third_party/libwebrtc/audio/voip/voip_core.h
new file mode 100644
index 0000000000..6c3aec6fa2
--- /dev/null
+++ b/third_party/libwebrtc/audio/voip/voip_core.h
@@ -0,0 +1,174 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_VOIP_VOIP_CORE_H_
+#define AUDIO_VOIP_VOIP_CORE_H_
+
+#include <map>
+#include <memory>
+#include <queue>
+#include <unordered_map>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/voip/voip_base.h"
+#include "api/voip/voip_codec.h"
+#include "api/voip/voip_dtmf.h"
+#include "api/voip/voip_engine.h"
+#include "api/voip/voip_network.h"
+#include "api/voip/voip_statistics.h"
+#include "api/voip/voip_volume_control.h"
+#include "audio/audio_transport_impl.h"
+#include "audio/voip/audio_channel.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// VoipCore is the implementatino of VoIP APIs listed in api/voip directory.
+// It manages a vector of AudioChannel objects where each is mapped with a
+// ChannelId (int) type. ChannelId is the primary key to locate a specific
+// AudioChannel object to operate requested VoIP API from the caller.
+//
+// This class receives required audio components from caller at construction and
+// owns the life cycle of them to orchestrate the proper destruction sequence.
+class VoipCore : public VoipEngine,
+                 public VoipBase,
+                 public VoipNetwork,
+                 public VoipCodec,
+                 public VoipDtmf,
+                 public VoipStatistics,
+                 public VoipVolumeControl {
+ public:
+  // Construct VoipCore with provided arguments.
+  VoipCore(rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
+           rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+           std::unique_ptr<TaskQueueFactory> task_queue_factory,
+           rtc::scoped_refptr<AudioDeviceModule> audio_device_module,
+           rtc::scoped_refptr<AudioProcessing> audio_processing);
+  ~VoipCore() override = default;
+
+  // Implements VoipEngine interfaces.
+  VoipBase& Base() override { return *this; }
+  VoipNetwork& Network() override { return *this; }
+  VoipCodec& Codec() override { return *this; }
+  VoipDtmf& Dtmf() override { return *this; }
+  VoipStatistics& Statistics() override { return *this; }
+  VoipVolumeControl& VolumeControl() override { return *this; }
+
+  // Implements VoipBase interfaces.
+  ChannelId CreateChannel(Transport* transport,
+                          absl::optional<uint32_t> local_ssrc) override;
+  VoipResult ReleaseChannel(ChannelId channel_id) override;
+  VoipResult StartSend(ChannelId channel_id) override;
+  VoipResult StopSend(ChannelId channel_id) override;
+  VoipResult StartPlayout(ChannelId channel_id) override;
+  VoipResult StopPlayout(ChannelId channel_id) override;
+
+  // Implements VoipNetwork interfaces.
+  VoipResult ReceivedRTPPacket(
+      ChannelId channel_id,
+      rtc::ArrayView<const uint8_t> rtp_packet) override;
+  VoipResult ReceivedRTCPPacket(
+      ChannelId channel_id,
+      rtc::ArrayView<const uint8_t> rtcp_packet) override;
+
+  // Implements VoipCodec interfaces.
+  VoipResult SetSendCodec(ChannelId channel_id,
+                          int payload_type,
+                          const SdpAudioFormat& encoder_format) override;
+  VoipResult SetReceiveCodecs(
+      ChannelId channel_id,
+      const std::map<int, SdpAudioFormat>& decoder_specs) override;
+
+  // Implements VoipDtmf interfaces.
+  VoipResult RegisterTelephoneEventType(ChannelId channel_id,
+                                        int rtp_payload_type,
+                                        int sample_rate_hz) override;
+  VoipResult SendDtmfEvent(ChannelId channel_id,
+                           DtmfEvent dtmf_event,
+                           int duration_ms) override;
+
+  // Implements VoipStatistics interfaces.
+  VoipResult GetIngressStatistics(ChannelId channel_id,
+                                  IngressStatistics& ingress_stats) override;
+  VoipResult GetChannelStatistics(ChannelId channe_id,
+                                  ChannelStatistics& channel_stats) override;
+
+  // Implements VoipVolumeControl interfaces.
+  VoipResult SetInputMuted(ChannelId channel_id, bool enable) override;
+  VoipResult GetInputVolumeInfo(ChannelId channel_id,
+                                VolumeInfo& volume_info) override;
+  VoipResult GetOutputVolumeInfo(ChannelId channel_id,
+                                 VolumeInfo& volume_info) override;
+
+ private:
+  // Initialize ADM and default audio device if needed.
+  // Returns true if ADM is successfully initialized or already in such state
+  // (e.g called more than once). Returns false when ADM fails to initialize
+  // which would presumably render further processing useless. Note that such
+  // failure won't necessarily succeed in next initialization attempt as it
+  // would mean changing the ADM implementation. From Android N and onwards, the
+  // mobile app may not be able to gain microphone access when in background
+  // mode. Therefore it would be better to delay the logic as late as possible.
+  bool InitializeIfNeeded();
+
+  // Fetches the corresponding AudioChannel assigned with given `channel`.
+  // Returns nullptr if not found.
+  rtc::scoped_refptr<AudioChannel> GetChannel(ChannelId channel_id);
+
+  // Updates AudioTransportImpl with a new set of actively sending AudioSender
+  // (AudioEgress). This needs to be invoked whenever StartSend/StopSend is
+  // involved by caller. Returns false when the selected audio device fails to
+  // initialize where it can't expect to deliver any audio input sample.
+  bool UpdateAudioTransportWithSenders();
+
+  // Synchronization for these are handled internally.
+  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+
+  // Synchronization is handled internally by AudioProcessing.
+  // Must be placed before `audio_device_module_` for proper destruction.
+  rtc::scoped_refptr<AudioProcessing> audio_processing_;
+
+  // Synchronization is handled internally by AudioMixer.
+  // Must be placed before `audio_device_module_` for proper destruction.
+  rtc::scoped_refptr<AudioMixer> audio_mixer_;
+
+  // Synchronization is handled internally by AudioTransportImpl.
+  // Must be placed before `audio_device_module_` for proper destruction.
+  std::unique_ptr<AudioTransportImpl> audio_transport_;
+
+  // Synchronization is handled internally by AudioDeviceModule.
+  rtc::scoped_refptr<AudioDeviceModule> audio_device_module_;
+
+  Mutex lock_;
+
+  // Member to track a next ChannelId for new AudioChannel.
+  int next_channel_id_ RTC_GUARDED_BY(lock_) = 0;
+
+  // Container to track currently active AudioChannel objects mapped by
+  // ChannelId.
+  std::unordered_map<ChannelId, rtc::scoped_refptr<AudioChannel>> channels_
+      RTC_GUARDED_BY(lock_);
+
+  // Boolean flag to ensure initialization only occurs once.
+  bool initialized_ RTC_GUARDED_BY(lock_) = false;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_VOIP_VOIP_CORE_H_