Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

80 files changed, 8955 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/audio_codecs/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/BUILD.gn
new file mode 100644
index 0000000000..82ed31a5da
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/BUILD.gn
@@ -0,0 +1,144 @@
+# 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.
+
+import("../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio_codecs_api") {
+  visibility = [ "*" ]
+  sources = [
+    "audio_codec_pair_id.cc",
+    "audio_codec_pair_id.h",
+    "audio_decoder.cc",
+    "audio_decoder.h",
+    "audio_decoder_factory.h",
+    "audio_decoder_factory_template.h",
+    "audio_encoder.cc",
+    "audio_encoder.h",
+    "audio_encoder_factory.h",
+    "audio_encoder_factory_template.h",
+    "audio_format.cc",
+    "audio_format.h",
+  ]
+  deps = [
+    "..:array_view",
+    "..:bitrate_allocation",
+    "..:make_ref_counted",
+    "..:scoped_refptr",
+    "../../api:field_trials_view",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:refcount",
+    "../../rtc_base:sanitizer",
+    "../../rtc_base/system:rtc_export",
+    "../units:time_delta",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("builtin_audio_decoder_factory") {
+  visibility = [ "*" ]
+  allow_poison = [ "audio_codecs" ]
+  sources = [
+    "builtin_audio_decoder_factory.cc",
+    "builtin_audio_decoder_factory.h",
+  ]
+  deps = [
+    ":audio_codecs_api",
+    "..:scoped_refptr",
+    "L16:audio_decoder_L16",
+    "g711:audio_decoder_g711",
+    "g722:audio_decoder_g722",
+  ]
+  defines = []
+  if (rtc_include_ilbc) {
+    deps += [ "ilbc:audio_decoder_ilbc" ]
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
+  }
+  if (rtc_include_opus) {
+    deps += [
+      "opus:audio_decoder_multiopus",
+      "opus:audio_decoder_opus",
+    ]
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
+  }
+}
+
+rtc_library("builtin_audio_encoder_factory") {
+  visibility = [ "*" ]
+  allow_poison = [ "audio_codecs" ]
+  sources = [
+    "builtin_audio_encoder_factory.cc",
+    "builtin_audio_encoder_factory.h",
+  ]
+  deps = [
+    ":audio_codecs_api",
+    "..:scoped_refptr",
+    "L16:audio_encoder_L16",
+    "g711:audio_encoder_g711",
+    "g722:audio_encoder_g722",
+  ]
+  defines = []
+  if (rtc_include_ilbc) {
+    deps += [ "ilbc:audio_encoder_ilbc" ]
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
+  }
+  if (rtc_include_opus) {
+    deps += [
+      "opus:audio_encoder_multiopus",
+      "opus:audio_encoder_opus",
+    ]
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
+  }
+}
+
+rtc_library("opus_audio_decoder_factory") {
+  visibility = [ "*" ]
+  allow_poison = [ "audio_codecs" ]
+  sources = [
+    "opus_audio_decoder_factory.cc",
+    "opus_audio_decoder_factory.h",
+  ]
+  deps = [
+    ":audio_codecs_api",
+    "..:scoped_refptr",
+    "opus:audio_decoder_multiopus",
+    "opus:audio_decoder_opus",
+  ]
+}
+
+rtc_library("opus_audio_encoder_factory") {
+  visibility = [ "*" ]
+  allow_poison = [ "audio_codecs" ]
+  sources = [
+    "opus_audio_encoder_factory.cc",
+    "opus_audio_encoder_factory.h",
+  ]
+  deps = [
+    ":audio_codecs_api",
+    "..:scoped_refptr",
+    "opus:audio_encoder_multiopus",
+    "opus:audio_encoder_opus",
+  ]
+}
diff --git a/third_party/libwebrtc/api/audio_codecs/L16/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/L16/BUILD.gn
new file mode 100644
index 0000000000..41e9eb42d8
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/L16/BUILD.gn
@@ -0,0 +1,55 @@
+# 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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio_encoder_L16") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_L16.cc",
+    "audio_encoder_L16.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:pcm16b",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_L16") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_L16.cc",
+    "audio_decoder_L16.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:pcm16b",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.cc b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.cc
new file mode 100644
index 0000000000..a03abe26f7
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.cc
@@ -0,0 +1,49 @@
+/*
+ *  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 "api/audio_codecs/L16/audio_decoder_L16.h"
+
+#include <memory>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
+    const SdpAudioFormat& format) {
+  Config config;
+  config.sample_rate_hz = format.clockrate_hz;
+  config.num_channels = rtc::checked_cast<int>(format.num_channels);
+  if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
+    return config;
+  }
+  return absl::nullopt;
+}
+
+void AudioDecoderL16::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  Pcm16BAppendSupportedCodecSpecs(specs);
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
+    const Config& config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    return nullptr;
+  }
+  return std::make_unique<AudioDecoderPcm16B>(config.sample_rate_hz,
+                                              config.num_channels);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.h b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.h
new file mode 100644
index 0000000000..5a01b7dc01
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.h
@@ -0,0 +1,49 @@
+/*
+ *  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 API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
+#define API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// L16 decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderL16 {
+  struct Config {
+    bool IsOk() const {
+      return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+              sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
+             (num_channels >= 1 &&
+              num_channels <= AudioDecoder::kMaxNumberOfChannels);
+    }
+    int sample_rate_hz = 8000;
+    int num_channels = 1;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config& config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_L16_AUDIO_DECODER_L16_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16_gn/moz.build
new file mode 100644
index 0000000000..87335c298d
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16_gn/moz.build
@@ -0,0 +1,225 @@
+# 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/api/audio_codecs/L16/audio_decoder_L16.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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_decoder_L16_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc
new file mode 100644
index 0000000000..20259b9ad8
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc
@@ -0,0 +1,76 @@
+/*
+ *  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 "api/audio_codecs/L16/audio_encoder_L16.h"
+
+#include <memory>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  Config config;
+  config.sample_rate_hz = format.clockrate_hz;
+  config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
+    }
+  }
+  if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
+    return config;
+  }
+  return absl::nullopt;
+}
+
+void AudioEncoderL16::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  Pcm16BAppendSupportedCodecSpecs(specs);
+}
+
+AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
+    const AudioEncoderL16::Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {config.sample_rate_hz,
+          rtc::dchecked_cast<size_t>(config.num_channels),
+          config.sample_rate_hz * config.num_channels * 16};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
+    const AudioEncoderL16::Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  AudioEncoderPcm16B::Config c;
+  c.sample_rate_hz = config.sample_rate_hz;
+  c.num_channels = config.num_channels;
+  c.frame_size_ms = config.frame_size_ms;
+  c.payload_type = payload_type;
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderPcm16B>(c);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h
new file mode 100644
index 0000000000..47509849de
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.h
@@ -0,0 +1,54 @@
+/*
+ *  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 API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
+#define API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// L16 encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderL16 {
+  struct Config {
+    bool IsOk() const {
+      return (sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+              sample_rate_hz == 32000 || sample_rate_hz == 48000) &&
+             num_channels >= 1 &&
+             num_channels <= AudioEncoder::kMaxNumberOfChannels &&
+             frame_size_ms > 0 && frame_size_ms <= 120 &&
+             frame_size_ms % 10 == 0;
+    }
+    int sample_rate_hz = 8000;
+    int num_channels = 1;
+    int frame_size_ms = 10;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_L16_AUDIO_ENCODER_L16_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16_gn/moz.build
new file mode 100644
index 0000000000..49e0d546f1
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16_gn/moz.build
@@ -0,0 +1,225 @@
+# 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/api/audio_codecs/L16/audio_encoder_L16.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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_encoder_L16_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/OWNERS b/third_party/libwebrtc/api/audio_codecs/OWNERS
new file mode 100644
index 0000000000..77b414abc3
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/OWNERS
@@ -0,0 +1,3 @@
+alessiob@webrtc.org
+henrik.lundin@webrtc.org
+jakobi@webrtc.org
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.cc b/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.cc
new file mode 100644
index 0000000000..6cb51ed6b7
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2018 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 "api/audio_codecs/audio_codec_pair_id.h"
+
+#include <atomic>
+#include <cstdint>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+// Returns a new value that it has never returned before. You may call it at
+// most 2^63 times in the lifetime of the program. Note: The returned values
+// may be easily predictable.
+uint64_t GetNextId() {
+  static std::atomic<uint64_t> next_id(0);
+
+  // Atomically increment `next_id`, and return the previous value. Relaxed
+  // memory order is sufficient, since all we care about is that different
+  // callers return different values.
+  const uint64_t new_id = next_id.fetch_add(1, std::memory_order_relaxed);
+
+  // This check isn't atomic with the increment, so if we start 2^63 + 1
+  // invocations of GetNextId() in parallel, the last one to do the atomic
+  // increment could return the ID 0 before any of the others had time to
+  // trigger this DCHECK. We blithely assume that this won't happen.
+  RTC_DCHECK_LT(new_id, uint64_t{1} << 63) << "Used up all ID values";
+
+  return new_id;
+}
+
+// Make an integer ID more unpredictable. This is a 1:1 mapping, so you can
+// feed it any value, but the idea is that you can feed it a sequence such as
+// 0, 1, 2, ... and get a new sequence that isn't as trivially predictable, so
+// that users won't rely on it being consecutive or increasing or anything like
+// that.
+constexpr uint64_t ObfuscateId(uint64_t id) {
+  // Any nonzero coefficient that's relatively prime to 2^64 (that is, any odd
+  // number) and any constant will give a 1:1 mapping. These high-entropy
+  // values will prevent the sequence from being trivially predictable.
+  //
+  // Both the multiplication and the addition going to overflow almost always,
+  // but that's fine---we *want* arithmetic mod 2^64.
+  return uint64_t{0x85fdb20e1294309a} + uint64_t{0xc516ef5c37462469} * id;
+}
+
+// The first ten values. Verified against the Python function
+//
+//   def f(n):
+//     return (0x85fdb20e1294309a + 0xc516ef5c37462469 * n) % 2**64
+//
+// Callers should obviously not depend on these exact values...
+//
+// (On Visual C++, we have to disable warning C4307 (integral constant
+// overflow), even though unsigned integers have perfectly well-defined
+// overflow behavior.)
+#ifdef _MSC_VER
+#pragma warning(push)
+#pragma warning(disable : 4307)
+#endif
+static_assert(ObfuscateId(0) == uint64_t{0x85fdb20e1294309a}, "");
+static_assert(ObfuscateId(1) == uint64_t{0x4b14a16a49da5503}, "");
+static_assert(ObfuscateId(2) == uint64_t{0x102b90c68120796c}, "");
+static_assert(ObfuscateId(3) == uint64_t{0xd5428022b8669dd5}, "");
+static_assert(ObfuscateId(4) == uint64_t{0x9a596f7eefacc23e}, "");
+static_assert(ObfuscateId(5) == uint64_t{0x5f705edb26f2e6a7}, "");
+static_assert(ObfuscateId(6) == uint64_t{0x24874e375e390b10}, "");
+static_assert(ObfuscateId(7) == uint64_t{0xe99e3d93957f2f79}, "");
+static_assert(ObfuscateId(8) == uint64_t{0xaeb52cefccc553e2}, "");
+static_assert(ObfuscateId(9) == uint64_t{0x73cc1c4c040b784b}, "");
+#ifdef _MSC_VER
+#pragma warning(pop)
+#endif
+
+}  // namespace
+
+AudioCodecPairId AudioCodecPairId::Create() {
+  return AudioCodecPairId(ObfuscateId(GetNextId()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.h b/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.h
new file mode 100644
index 0000000000..b10f14ea66
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_codec_pair_id.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2018 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 API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_
+#define API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_
+
+#include <stdint.h>
+
+#include <utility>
+
+namespace webrtc {
+
+class AudioCodecPairId final {
+ public:
+  // Copyable, but not default constructible.
+  AudioCodecPairId() = delete;
+  AudioCodecPairId(const AudioCodecPairId&) = default;
+  AudioCodecPairId(AudioCodecPairId&&) = default;
+  AudioCodecPairId& operator=(const AudioCodecPairId&) = default;
+  AudioCodecPairId& operator=(AudioCodecPairId&&) = default;
+
+  friend void swap(AudioCodecPairId& a, AudioCodecPairId& b) {
+    using std::swap;
+    swap(a.id_, b.id_);
+  }
+
+  // Creates a new ID, unequal to any previously created ID.
+  static AudioCodecPairId Create();
+
+  // IDs can be tested for equality.
+  friend bool operator==(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ == b.id_;
+  }
+  friend bool operator!=(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ != b.id_;
+  }
+
+  // Comparisons. The ordering of ID values is completely arbitrary, but
+  // stable, so it's useful e.g. if you want to use IDs as keys in an ordered
+  // map.
+  friend bool operator<(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ < b.id_;
+  }
+  friend bool operator<=(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ <= b.id_;
+  }
+  friend bool operator>=(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ >= b.id_;
+  }
+  friend bool operator>(AudioCodecPairId a, AudioCodecPairId b) {
+    return a.id_ > b.id_;
+  }
+
+  // Returns a numeric representation of the ID. The numeric values are
+  // completely arbitrary, but stable, collision-free, and reasonably evenly
+  // distributed, so they are e.g. useful as hash values in unordered maps.
+  uint64_t NumericRepresentation() const { return id_; }
+
+ private:
+  explicit AudioCodecPairId(uint64_t id) : id_(id) {}
+
+  uint64_t id_;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_codecs_api_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/audio_codecs_api_gn/moz.build
new file mode 100644
index 0000000000..846946073e
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_codecs_api_gn/moz.build
@@ -0,0 +1,228 @@
+# 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/api/audio_codecs/audio_codec_pair_id.cc",
+    "/third_party/libwebrtc/api/audio_codecs/audio_decoder.cc",
+    "/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc",
+    "/third_party/libwebrtc/api/audio_codecs/audio_format.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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_codecs_api_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_decoder.cc b/third_party/libwebrtc/api/audio_codecs/audio_decoder.cc
new file mode 100644
index 0000000000..28f5b8aae8
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_decoder.cc
@@ -0,0 +1,170 @@
+/*
+ *  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 "api/audio_codecs/audio_decoder.h"
+
+
+#include <memory>
+#include <utility>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/sanitizer.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+namespace {
+
+class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame {
+ public:
+  OldStyleEncodedFrame(AudioDecoder* decoder, rtc::Buffer&& payload)
+      : decoder_(decoder), payload_(std::move(payload)) {}
+
+  size_t Duration() const override {
+    const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    return ret < 0 ? 0 : static_cast<size_t>(ret);
+  }
+
+  absl::optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    auto speech_type = AudioDecoder::kSpeech;
+    const int ret = decoder_->Decode(
+        payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+        decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    return ret < 0 ? absl::nullopt
+                   : absl::optional<DecodeResult>(
+                         {static_cast<size_t>(ret), speech_type});
+  }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+};
+
+}  // namespace
+
+bool AudioDecoder::EncodedAudioFrame::IsDtxPacket() const {
+  return false;
+}
+
+AudioDecoder::ParseResult::ParseResult() = default;
+AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default;
+AudioDecoder::ParseResult::ParseResult(uint32_t timestamp,
+                                       int priority,
+                                       std::unique_ptr<EncodedAudioFrame> frame)
+    : timestamp(timestamp), priority(priority), frame(std::move(frame)) {
+  RTC_DCHECK_GE(priority, 0);
+}
+
+AudioDecoder::ParseResult::~ParseResult() = default;
+
+AudioDecoder::ParseResult& AudioDecoder::ParseResult::operator=(
+    ParseResult&& b) = default;
+
+std::vector<AudioDecoder::ParseResult> AudioDecoder::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OldStyleEncodedFrame(this, std::move(payload)));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int AudioDecoder::Decode(const uint8_t* encoded,
+                         size_t encoded_len,
+                         int sample_rate_hz,
+                         size_t max_decoded_bytes,
+                         int16_t* decoded,
+                         SpeechType* speech_type) {
+  TRACE_EVENT0("webrtc", "AudioDecoder::Decode");
+  rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
+  int duration = PacketDuration(encoded, encoded_len);
+  if (duration >= 0 &&
+      duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
+    return -1;
+  }
+  return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                        speech_type);
+}
+
+int AudioDecoder::DecodeRedundant(const uint8_t* encoded,
+                                  size_t encoded_len,
+                                  int sample_rate_hz,
+                                  size_t max_decoded_bytes,
+                                  int16_t* decoded,
+                                  SpeechType* speech_type) {
+  TRACE_EVENT0("webrtc", "AudioDecoder::DecodeRedundant");
+  rtc::MsanCheckInitialized(rtc::MakeArrayView(encoded, encoded_len));
+  int duration = PacketDurationRedundant(encoded, encoded_len);
+  if (duration >= 0 &&
+      duration * Channels() * sizeof(int16_t) > max_decoded_bytes) {
+    return -1;
+  }
+  return DecodeRedundantInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                                 speech_type);
+}
+
+int AudioDecoder::DecodeRedundantInternal(const uint8_t* encoded,
+                                          size_t encoded_len,
+                                          int sample_rate_hz,
+                                          int16_t* decoded,
+                                          SpeechType* speech_type) {
+  return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                        speech_type);
+}
+
+bool AudioDecoder::HasDecodePlc() const {
+  return false;
+}
+
+size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return 0;
+}
+
+// TODO(bugs.webrtc.org/9676): Remove default implementation.
+void AudioDecoder::GeneratePlc(size_t /*requested_samples_per_channel*/,
+                               rtc::BufferT<int16_t>* /*concealment_audio*/) {}
+
+int AudioDecoder::ErrorCode() {
+  return 0;
+}
+
+int AudioDecoder::PacketDuration(const uint8_t* encoded,
+                                 size_t encoded_len) const {
+  return kNotImplemented;
+}
+
+int AudioDecoder::PacketDurationRedundant(const uint8_t* encoded,
+                                          size_t encoded_len) const {
+  return kNotImplemented;
+}
+
+bool AudioDecoder::PacketHasFec(const uint8_t* encoded,
+                                size_t encoded_len) const {
+  return false;
+}
+
+AudioDecoder::SpeechType AudioDecoder::ConvertSpeechType(int16_t type) {
+  switch (type) {
+    case 0:  // TODO(hlundin): Both iSAC and Opus return 0 for speech.
+    case 1:
+      return kSpeech;
+    case 2:
+      return kComfortNoise;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return kSpeech;
+  }
+}
+
+constexpr int AudioDecoder::kMaxNumberOfChannels;
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_decoder.h b/third_party/libwebrtc/api/audio_codecs/audio_decoder.h
new file mode 100644
index 0000000000..41138741bb
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_decoder.h
@@ -0,0 +1,195 @@
+/*
+ *  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 API_AUDIO_CODECS_AUDIO_DECODER_H_
+#define API_AUDIO_CODECS_AUDIO_DECODER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoder {
+ public:
+  enum SpeechType {
+    kSpeech = 1,
+    kComfortNoise = 2,
+  };
+
+  // Used by PacketDuration below. Save the value -1 for errors.
+  enum { kNotImplemented = -2 };
+
+  AudioDecoder() = default;
+  virtual ~AudioDecoder() = default;
+
+  AudioDecoder(const AudioDecoder&) = delete;
+  AudioDecoder& operator=(const AudioDecoder&) = delete;
+
+  class EncodedAudioFrame {
+   public:
+    struct DecodeResult {
+      size_t num_decoded_samples;
+      SpeechType speech_type;
+    };
+
+    virtual ~EncodedAudioFrame() = default;
+
+    // Returns the duration in samples-per-channel of this audio frame.
+    // If no duration can be ascertained, returns zero.
+    virtual size_t Duration() const = 0;
+
+    // Returns true if this packet contains DTX.
+    virtual bool IsDtxPacket() const;
+
+    // Decodes this frame of audio and writes the result in `decoded`.
+    // `decoded` must be large enough to store as many samples as indicated by a
+    // call to Duration() . On success, returns an absl::optional containing the
+    // total number of samples across all channels, as well as whether the
+    // decoder produced comfort noise or speech. On failure, returns an empty
+    // absl::optional. Decode may be called at most once per frame object.
+    virtual absl::optional<DecodeResult> Decode(
+        rtc::ArrayView<int16_t> decoded) const = 0;
+  };
+
+  struct ParseResult {
+    ParseResult();
+    ParseResult(uint32_t timestamp,
+                int priority,
+                std::unique_ptr<EncodedAudioFrame> frame);
+    ParseResult(ParseResult&& b);
+    ~ParseResult();
+
+    ParseResult& operator=(ParseResult&& b);
+
+    // The timestamp of the frame is in samples per channel.
+    uint32_t timestamp;
+    // The relative priority of the frame compared to other frames of the same
+    // payload and the same timeframe. A higher value means a lower priority.
+    // The highest priority is zero - negative values are not allowed.
+    int priority;
+    std::unique_ptr<EncodedAudioFrame> frame;
+  };
+
+  // Let the decoder parse this payload and prepare zero or more decodable
+  // frames. Each frame must be between 10 ms and 120 ms long. The caller must
+  // ensure that the AudioDecoder object outlives any frame objects returned by
+  // this call. The decoder is free to swap or move the data from the `payload`
+  // buffer. `timestamp` is the input timestamp, in samples, corresponding to
+  // the start of the payload.
+  virtual std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                                uint32_t timestamp);
+
+  // TODO(bugs.webrtc.org/10098): The Decode and DecodeRedundant methods are
+  // obsolete; callers should call ParsePayload instead. For now, subclasses
+  // must still implement DecodeInternal.
+
+  // Decodes `encode_len` bytes from `encoded` and writes the result in
+  // `decoded`. The maximum bytes allowed to be written into `decoded` is
+  // `max_decoded_bytes`. Returns the total number of samples across all
+  // channels. If the decoder produced comfort noise, `speech_type`
+  // is set to kComfortNoise, otherwise it is kSpeech. The desired output
+  // sample rate is provided in `sample_rate_hz`, which must be valid for the
+  // codec at hand.
+  int Decode(const uint8_t* encoded,
+             size_t encoded_len,
+             int sample_rate_hz,
+             size_t max_decoded_bytes,
+             int16_t* decoded,
+             SpeechType* speech_type);
+
+  // Same as Decode(), but interfaces to the decoders redundant decode function.
+  // The default implementation simply calls the regular Decode() method.
+  int DecodeRedundant(const uint8_t* encoded,
+                      size_t encoded_len,
+                      int sample_rate_hz,
+                      size_t max_decoded_bytes,
+                      int16_t* decoded,
+                      SpeechType* speech_type);
+
+  // Indicates if the decoder implements the DecodePlc method.
+  virtual bool HasDecodePlc() const;
+
+  // Calls the packet-loss concealment of the decoder to update the state after
+  // one or several lost packets. The caller has to make sure that the
+  // memory allocated in `decoded` should accommodate `num_frames` frames.
+  virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
+
+  // Asks the decoder to generate packet-loss concealment and append it to the
+  // end of `concealment_audio`. The concealment audio should be in
+  // channel-interleaved format, with as many channels as the last decoded
+  // packet produced. The implementation must produce at least
+  // requested_samples_per_channel, or nothing at all. This is a signal to the
+  // caller to conceal the loss with other means. If the implementation provides
+  // concealment samples, it is also responsible for "stitching" it together
+  // with the decoded audio on either side of the concealment.
+  // Note: The default implementation of GeneratePlc will be deleted soon. All
+  // implementations must provide their own, which can be a simple as a no-op.
+  // TODO(bugs.webrtc.org/9676): Remove default implementation.
+  virtual void GeneratePlc(size_t requested_samples_per_channel,
+                           rtc::BufferT<int16_t>* concealment_audio);
+
+  // Resets the decoder state (empty buffers etc.).
+  virtual void Reset() = 0;
+
+  // Returns the last error code from the decoder.
+  virtual int ErrorCode();
+
+  // Returns the duration in samples-per-channel of the payload in `encoded`
+  // which is `encoded_len` bytes long. Returns kNotImplemented if no duration
+  // estimate is available, or -1 in case of an error.
+  virtual int PacketDuration(const uint8_t* encoded, size_t encoded_len) const;
+
+  // Returns the duration in samples-per-channel of the redandant payload in
+  // `encoded` which is `encoded_len` bytes long. Returns kNotImplemented if no
+  // duration estimate is available, or -1 in case of an error.
+  virtual int PacketDurationRedundant(const uint8_t* encoded,
+                                      size_t encoded_len) const;
+
+  // Detects whether a packet has forward error correction. The packet is
+  // comprised of the samples in `encoded` which is `encoded_len` bytes long.
+  // Returns true if the packet has FEC and false otherwise.
+  virtual bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const;
+
+  // Returns the actual sample rate of the decoder's output. This value may not
+  // change during the lifetime of the decoder.
+  virtual int SampleRateHz() const = 0;
+
+  // The number of channels in the decoder's output. This value may not change
+  // during the lifetime of the decoder.
+  virtual size_t Channels() const = 0;
+
+  // The maximum number of audio channels supported by WebRTC decoders.
+  static constexpr int kMaxNumberOfChannels = 24;
+
+ protected:
+  static SpeechType ConvertSpeechType(int16_t type);
+
+  virtual int DecodeInternal(const uint8_t* encoded,
+                             size_t encoded_len,
+                             int sample_rate_hz,
+                             int16_t* decoded,
+                             SpeechType* speech_type) = 0;
+
+  virtual int DecodeRedundantInternal(const uint8_t* encoded,
+                                      size_t encoded_len,
+                                      int sample_rate_hz,
+                                      int16_t* decoded,
+                                      SpeechType* speech_type);
+};
+
+}  // namespace webrtc
+#endif  // API_AUDIO_CODECS_AUDIO_DECODER_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory.h b/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory.h
new file mode 100644
index 0000000000..2811f6704b
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory.h
@@ -0,0 +1,53 @@
+/*
+ *  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 API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
+#define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// A factory that creates AudioDecoders.
+class AudioDecoderFactory : public rtc::RefCountInterface {
+ public:
+  virtual std::vector<AudioCodecSpec> GetSupportedDecoders() = 0;
+
+  virtual bool IsSupportedDecoder(const SdpAudioFormat& format) = 0;
+
+  // Create a new decoder instance. The `codec_pair_id` argument is used to link
+  // encoders and decoders that talk to the same remote entity: if a
+  // AudioEncoderFactory::MakeAudioEncoder() and a
+  // AudioDecoderFactory::MakeAudioDecoder() call receive non-null IDs that
+  // compare equal, the factory implementations may assume that the encoder and
+  // decoder form a pair. (The intended use case for this is to set up
+  // communication between the AudioEncoder and AudioDecoder instances, which is
+  // needed for some codecs with built-in bandwidth adaptation.)
+  //
+  // Returns null if the format isn't supported.
+  //
+  // Note: Implementations need to be robust against combinations other than
+  // one encoder, one decoder getting the same ID; such decoders must still
+  // work.
+  virtual std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory_template.h b/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory_template.h
new file mode 100644
index 0000000000..7ea0c91372
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory_template.h
@@ -0,0 +1,145 @@
+/*
+ *  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 API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
+#define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/field_trials_view.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+namespace audio_decoder_factory_template_impl {
+
+template <typename... Ts>
+struct Helper;
+
+// Base case: 0 template parameters.
+template <>
+struct Helper<> {
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {}
+  static bool IsSupportedDecoder(const SdpAudioFormat& format) { return false; }
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const FieldTrialsView* field_trials) {
+    return nullptr;
+  }
+};
+
+// Inductive case: Called with n + 1 template parameters; calls subroutines
+// with n template parameters.
+template <typename T, typename... Ts>
+struct Helper<T, Ts...> {
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
+    T::AppendSupportedDecoders(specs);
+    Helper<Ts...>::AppendSupportedDecoders(specs);
+  }
+  static bool IsSupportedDecoder(const SdpAudioFormat& format) {
+    auto opt_config = T::SdpToConfig(format);
+    static_assert(std::is_same<decltype(opt_config),
+                               absl::optional<typename T::Config>>::value,
+                  "T::SdpToConfig() must return a value of type "
+                  "absl::optional<T::Config>");
+    return opt_config ? true : Helper<Ts...>::IsSupportedDecoder(format);
+  }
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const FieldTrialsView* field_trials) {
+    auto opt_config = T::SdpToConfig(format);
+    return opt_config ? T::MakeAudioDecoder(*opt_config, codec_pair_id)
+                      : Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id,
+                                                        field_trials);
+  }
+};
+
+template <typename... Ts>
+class AudioDecoderFactoryT : public AudioDecoderFactory {
+ public:
+  explicit AudioDecoderFactoryT(const FieldTrialsView* field_trials) {
+    field_trials_ = field_trials;
+  }
+
+  std::vector<AudioCodecSpec> GetSupportedDecoders() override {
+    std::vector<AudioCodecSpec> specs;
+    Helper<Ts...>::AppendSupportedDecoders(&specs);
+    return specs;
+  }
+
+  bool IsSupportedDecoder(const SdpAudioFormat& format) override {
+    return Helper<Ts...>::IsSupportedDecoder(format);
+  }
+
+  std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) override {
+    return Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id,
+                                           field_trials_);
+  }
+
+  const FieldTrialsView* field_trials_;
+};
+
+}  // namespace audio_decoder_factory_template_impl
+
+// Make an AudioDecoderFactory that can create instances of the given decoders.
+//
+// Each decoder type is given as a template argument to the function; it should
+// be a struct with the following static member functions:
+//
+//   // Converts `audio_format` to a ConfigType instance. Returns an empty
+//   // optional if `audio_format` doesn't correctly specify a decoder of our
+//   // type.
+//   absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
+//
+//   // Appends zero or more AudioCodecSpecs to the list that will be returned
+//   // by AudioDecoderFactory::GetSupportedDecoders().
+//   void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+//
+//   // Creates an AudioDecoder for the specified format. Used to implement
+//   // AudioDecoderFactory::MakeAudioDecoder().
+//   std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+//       const ConfigType& config,
+//       absl::optional<AudioCodecPairId> codec_pair_id);
+//
+// ConfigType should be a type that encapsulates all the settings needed to
+// create an AudioDecoder. T::Config (where T is the decoder struct) should
+// either be the config type, or an alias for it.
+//
+// Whenever it tries to do something, the new factory will try each of the
+// decoder types in the order they were specified in the template argument
+// list, stopping at the first one that claims to be able to do the job.
+//
+// TODO(kwiberg): Point at CreateBuiltinAudioDecoderFactory() for an example of
+// how it is used.
+template <typename... Ts>
+rtc::scoped_refptr<AudioDecoderFactory> CreateAudioDecoderFactory(
+    const FieldTrialsView* field_trials = nullptr) {
+  // There's no technical reason we couldn't allow zero template parameters,
+  // but such a factory couldn't create any decoders, and callers can do this
+  // by mistake by simply forgetting the <> altogether. So we forbid it in
+  // order to prevent caller foot-shooting.
+  static_assert(sizeof...(Ts) >= 1,
+                "Caller must give at least one template parameter");
+
+  return rtc::make_ref_counted<
+      audio_decoder_factory_template_impl::AudioDecoderFactoryT<Ts...>>(
+      field_trials);
+}
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc b/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc
new file mode 100644
index 0000000000..31bb8739f7
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2014 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 "api/audio_codecs/audio_encoder.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+ANAStats::ANAStats() = default;
+ANAStats::~ANAStats() = default;
+ANAStats::ANAStats(const ANAStats&) = default;
+
+AudioEncoder::EncodedInfo::EncodedInfo() = default;
+AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
+AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
+AudioEncoder::EncodedInfo::~EncodedInfo() = default;
+AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
+    const EncodedInfo&) = default;
+AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
+    default;
+
+int AudioEncoder::RtpTimestampRateHz() const {
+  return SampleRateHz();
+}
+
+AudioEncoder::EncodedInfo AudioEncoder::Encode(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
+  RTC_CHECK_EQ(audio.size(),
+               static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
+
+  const size_t old_size = encoded->size();
+  EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
+  RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
+  return info;
+}
+
+bool AudioEncoder::SetFec(bool enable) {
+  return !enable;
+}
+
+bool AudioEncoder::SetDtx(bool enable) {
+  return !enable;
+}
+
+bool AudioEncoder::GetDtx() const {
+  return false;
+}
+
+bool AudioEncoder::SetApplication(Application application) {
+  return false;
+}
+
+void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
+
+void AudioEncoder::SetTargetBitrate(int target_bps) {}
+
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoder::ReclaimContainedEncoders() {
+  return nullptr;
+}
+
+bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
+                                             RtcEventLog* event_log) {
+  return false;
+}
+
+void AudioEncoder::DisableAudioNetworkAdaptor() {}
+
+void AudioEncoder::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {}
+
+void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
+    float uplink_recoverable_packet_loss_fraction) {
+  RTC_DCHECK_NOTREACHED();
+}
+
+void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
+  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt);
+}
+
+void AudioEncoder::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {}
+
+void AudioEncoder::OnReceivedUplinkAllocation(BitrateAllocationUpdate update) {
+  OnReceivedUplinkBandwidth(update.target_bitrate.bps(),
+                            update.bwe_period.ms());
+}
+
+void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
+
+void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
+
+void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                               int max_frame_length_ms) {}
+
+ANAStats AudioEncoder::GetANAStats() const {
+  return ANAStats();
+}
+
+constexpr int AudioEncoder::kMaxNumberOfChannels;
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder.h b/third_party/libwebrtc/api/audio_codecs/audio_encoder.h
new file mode 100644
index 0000000000..7f5a34214f
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder.h
@@ -0,0 +1,260 @@
+/*
+ *  Copyright (c) 2014 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 API_AUDIO_CODECS_AUDIO_ENCODER_H_
+#define API_AUDIO_CODECS_AUDIO_ENCODER_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/call/bitrate_allocation.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+// Statistics related to Audio Network Adaptation.
+struct ANAStats {
+  ANAStats();
+  ANAStats(const ANAStats&);
+  ~ANAStats();
+  // Number of actions taken by the ANA bitrate controller since the start of
+  // the call. If this value is not set, it indicates that the bitrate
+  // controller is disabled.
+  absl::optional<uint32_t> bitrate_action_counter;
+  // Number of actions taken by the ANA channel controller since the start of
+  // the call. If this value is not set, it indicates that the channel
+  // controller is disabled.
+  absl::optional<uint32_t> channel_action_counter;
+  // Number of actions taken by the ANA DTX controller since the start of the
+  // call. If this value is not set, it indicates that the DTX controller is
+  // disabled.
+  absl::optional<uint32_t> dtx_action_counter;
+  // Number of actions taken by the ANA FEC controller since the start of the
+  // call. If this value is not set, it indicates that the FEC controller is
+  // disabled.
+  absl::optional<uint32_t> fec_action_counter;
+  // Number of times the ANA frame length controller decided to increase the
+  // frame length since the start of the call. If this value is not set, it
+  // indicates that the frame length controller is disabled.
+  absl::optional<uint32_t> frame_length_increase_counter;
+  // Number of times the ANA frame length controller decided to decrease the
+  // frame length since the start of the call. If this value is not set, it
+  // indicates that the frame length controller is disabled.
+  absl::optional<uint32_t> frame_length_decrease_counter;
+  // The uplink packet loss fractions as set by the ANA FEC controller. If this
+  // value is not set, it indicates that the ANA FEC controller is not active.
+  absl::optional<float> uplink_packet_loss_fraction;
+};
+
+// This is the interface class for encoders in AudioCoding module. Each codec
+// type must have an implementation of this class.
+class AudioEncoder {
+ public:
+  // Used for UMA logging of codec usage. The same codecs, with the
+  // same values, must be listed in
+  // src/tools/metrics/histograms/histograms.xml in chromium to log
+  // correct values.
+  enum class CodecType {
+    kOther = 0,  // Codec not specified, and/or not listed in this enum
+    kOpus = 1,
+    kIsac = 2,
+    kPcmA = 3,
+    kPcmU = 4,
+    kG722 = 5,
+    kIlbc = 6,
+
+    // Number of histogram bins in the UMA logging of codec types. The
+    // total number of different codecs that are logged cannot exceed this
+    // number.
+    kMaxLoggedAudioCodecTypes
+  };
+
+  struct EncodedInfoLeaf {
+    size_t encoded_bytes = 0;
+    uint32_t encoded_timestamp = 0;
+    int payload_type = 0;
+    bool send_even_if_empty = false;
+    bool speech = true;
+    CodecType encoder_type = CodecType::kOther;
+  };
+
+  // This is the main struct for auxiliary encoding information. Each encoded
+  // packet should be accompanied by one EncodedInfo struct, containing the
+  // total number of `encoded_bytes`, the `encoded_timestamp` and the
+  // `payload_type`. If the packet contains redundant encodings, the `redundant`
+  // vector will be populated with EncodedInfoLeaf structs. Each struct in the
+  // vector represents one encoding; the order of structs in the vector is the
+  // same as the order in which the actual payloads are written to the byte
+  // stream. When EncoderInfoLeaf structs are present in the vector, the main
+  // struct's `encoded_bytes` will be the sum of all the `encoded_bytes` in the
+  // vector.
+  struct EncodedInfo : public EncodedInfoLeaf {
+    EncodedInfo();
+    EncodedInfo(const EncodedInfo&);
+    EncodedInfo(EncodedInfo&&);
+    ~EncodedInfo();
+    EncodedInfo& operator=(const EncodedInfo&);
+    EncodedInfo& operator=(EncodedInfo&&);
+
+    std::vector<EncodedInfoLeaf> redundant;
+  };
+
+  virtual ~AudioEncoder() = default;
+
+  // Returns the input sample rate in Hz and the number of input channels.
+  // These are constants set at instantiation time.
+  virtual int SampleRateHz() const = 0;
+  virtual size_t NumChannels() const = 0;
+
+  // Returns the rate at which the RTP timestamps are updated. The default
+  // implementation returns SampleRateHz().
+  virtual int RtpTimestampRateHz() const;
+
+  // Returns the number of 10 ms frames the encoder will put in the next
+  // packet. This value may only change when Encode() outputs a packet; i.e.,
+  // the encoder may vary the number of 10 ms frames from packet to packet, but
+  // it must decide the length of the next packet no later than when outputting
+  // the preceding packet.
+  virtual size_t Num10MsFramesInNextPacket() const = 0;
+
+  // Returns the maximum value that can be returned by
+  // Num10MsFramesInNextPacket().
+  virtual size_t Max10MsFramesInAPacket() const = 0;
+
+  // Returns the current target bitrate in bits/s. The value -1 means that the
+  // codec adapts the target automatically, and a current target cannot be
+  // provided.
+  virtual int GetTargetBitrate() const = 0;
+
+  // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
+  // NumChannels() samples). Multi-channel audio must be sample-interleaved.
+  // The encoder appends zero or more bytes of output to `encoded` and returns
+  // additional encoding information.  Encode() checks some preconditions, calls
+  // EncodeImpl() which does the actual work, and then checks some
+  // postconditions.
+  EncodedInfo Encode(uint32_t rtp_timestamp,
+                     rtc::ArrayView<const int16_t> audio,
+                     rtc::Buffer* encoded);
+
+  // Resets the encoder to its starting state, discarding any input that has
+  // been fed to the encoder but not yet emitted in a packet.
+  virtual void Reset() = 0;
+
+  // Enables or disables codec-internal FEC (forward error correction). Returns
+  // true if the codec was able to comply. The default implementation returns
+  // true when asked to disable FEC and false when asked to enable it (meaning
+  // that FEC isn't supported).
+  virtual bool SetFec(bool enable);
+
+  // Enables or disables codec-internal VAD/DTX. Returns true if the codec was
+  // able to comply. The default implementation returns true when asked to
+  // disable DTX and false when asked to enable it (meaning that DTX isn't
+  // supported).
+  virtual bool SetDtx(bool enable);
+
+  // Returns the status of codec-internal DTX. The default implementation always
+  // returns false.
+  virtual bool GetDtx() const;
+
+  // Sets the application mode. Returns true if the codec was able to comply.
+  // The default implementation just returns false.
+  enum class Application { kSpeech, kAudio };
+  virtual bool SetApplication(Application application);
+
+  // Tells the encoder about the highest sample rate the decoder is expected to
+  // use when decoding the bitstream. The encoder would typically use this
+  // information to adjust the quality of the encoding. The default
+  // implementation does nothing.
+  virtual void SetMaxPlaybackRate(int frequency_hz);
+
+  // Tells the encoder what average bitrate we'd like it to produce. The
+  // encoder is free to adjust or disregard the given bitrate (the default
+  // implementation does the latter).
+  ABSL_DEPRECATED("Use OnReceivedTargetAudioBitrate instead")
+  virtual void SetTargetBitrate(int target_bps);
+
+  // Causes this encoder to let go of any other encoders it contains, and
+  // returns a pointer to an array where they are stored (which is required to
+  // live as long as this encoder). Unless the returned array is empty, you may
+  // not call any methods on this encoder afterwards, except for the
+  // destructor. The default implementation just returns an empty array.
+  // NOTE: This method is subject to change. Do not call or override it.
+  virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+  ReclaimContainedEncoders();
+
+  // Enables audio network adaptor. Returns true if successful.
+  virtual bool EnableAudioNetworkAdaptor(const std::string& config_string,
+                                         RtcEventLog* event_log);
+
+  // Disables audio network adaptor.
+  virtual void DisableAudioNetworkAdaptor();
+
+  // Provides uplink packet loss fraction to this encoder to allow it to adapt.
+  // `uplink_packet_loss_fraction` is in the range [0.0, 1.0].
+  virtual void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction);
+
+  ABSL_DEPRECATED("")
+  virtual void OnReceivedUplinkRecoverablePacketLossFraction(
+      float uplink_recoverable_packet_loss_fraction);
+
+  // Provides target audio bitrate to this encoder to allow it to adapt.
+  virtual void OnReceivedTargetAudioBitrate(int target_bps);
+
+  // Provides target audio bitrate and corresponding probing interval of
+  // the bandwidth estimator to this encoder to allow it to adapt.
+  virtual void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
+                                         absl::optional<int64_t> bwe_period_ms);
+
+  // Provides target audio bitrate and corresponding probing interval of
+  // the bandwidth estimator to this encoder to allow it to adapt.
+  virtual void OnReceivedUplinkAllocation(BitrateAllocationUpdate update);
+
+  // Provides RTT to this encoder to allow it to adapt.
+  virtual void OnReceivedRtt(int rtt_ms);
+
+  // Provides overhead to this encoder to adapt. The overhead is the number of
+  // bytes that will be added to each packet the encoder generates.
+  virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet);
+
+  // To allow encoder to adapt its frame length, it must be provided the frame
+  // length range that receivers can accept.
+  virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                           int max_frame_length_ms);
+
+  // Get statistics related to audio network adaptation.
+  virtual ANAStats GetANAStats() const;
+
+  // The range of frame lengths that are supported or nullopt if there's no sch
+  // information. This is used to calculated the full bitrate range, including
+  // overhead.
+  virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const = 0;
+
+  // The maximum number of audio channels supported by WebRTC encoders.
+  static constexpr int kMaxNumberOfChannels = 24;
+
+ protected:
+  // Subclasses implement this to perform the actual encoding. Called by
+  // Encode().
+  virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                                 rtc::ArrayView<const int16_t> audio,
+                                 rtc::Buffer* encoded) = 0;
+};
+}  // namespace webrtc
+#endif  // API_AUDIO_CODECS_AUDIO_ENCODER_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory.h b/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory.h
new file mode 100644
index 0000000000..6128b1b6f3
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory.h
@@ -0,0 +1,62 @@
+/*
+ *  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 API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
+#define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "rtc_base/ref_count.h"
+
+namespace webrtc {
+
+// A factory that creates AudioEncoders.
+class AudioEncoderFactory : public rtc::RefCountInterface {
+ public:
+  // Returns a prioritized list of audio codecs, to use for signaling etc.
+  virtual std::vector<AudioCodecSpec> GetSupportedEncoders() = 0;
+
+  // Returns information about how this format would be encoded, provided it's
+  // supported. More format and format variations may be supported than those
+  // returned by GetSupportedEncoders().
+  virtual absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format) = 0;
+
+  // Creates an AudioEncoder for the specified format. The encoder will tags its
+  // payloads with the specified payload type. The `codec_pair_id` argument is
+  // used to link encoders and decoders that talk to the same remote entity: if
+  // a AudioEncoderFactory::MakeAudioEncoder() and a
+  // AudioDecoderFactory::MakeAudioDecoder() call receive non-null IDs that
+  // compare equal, the factory implementations may assume that the encoder and
+  // decoder form a pair. (The intended use case for this is to set up
+  // communication between the AudioEncoder and AudioDecoder instances, which is
+  // needed for some codecs with built-in bandwidth adaptation.)
+  //
+  // Returns null if the format isn't supported.
+  //
+  // Note: Implementations need to be robust against combinations other than
+  // one encoder, one decoder getting the same ID; such encoders must still
+  // work.
+  //
+  // TODO(ossu): Try to avoid audio encoders having to know their payload type.
+  virtual std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      int payload_type,
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory_template.h b/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory_template.h
new file mode 100644
index 0000000000..8a70ba2268
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory_template.h
@@ -0,0 +1,163 @@
+/*
+ *  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 API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
+#define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/field_trials_view.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+namespace audio_encoder_factory_template_impl {
+
+template <typename... Ts>
+struct Helper;
+
+// Base case: 0 template parameters.
+template <>
+struct Helper<> {
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {}
+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format) {
+    return absl::nullopt;
+  }
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      int payload_type,
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const FieldTrialsView* field_trials) {
+    return nullptr;
+  }
+};
+
+// Inductive case: Called with n + 1 template parameters; calls subroutines
+// with n template parameters.
+template <typename T, typename... Ts>
+struct Helper<T, Ts...> {
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
+    T::AppendSupportedEncoders(specs);
+    Helper<Ts...>::AppendSupportedEncoders(specs);
+  }
+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format) {
+    auto opt_config = T::SdpToConfig(format);
+    static_assert(std::is_same<decltype(opt_config),
+                               absl::optional<typename T::Config>>::value,
+                  "T::SdpToConfig() must return a value of type "
+                  "absl::optional<T::Config>");
+    return opt_config ? absl::optional<AudioCodecInfo>(
+                            T::QueryAudioEncoder(*opt_config))
+                      : Helper<Ts...>::QueryAudioEncoder(format);
+  }
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      int payload_type,
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id,
+      const FieldTrialsView* field_trials) {
+    auto opt_config = T::SdpToConfig(format);
+    if (opt_config) {
+      return T::MakeAudioEncoder(*opt_config, payload_type, codec_pair_id);
+    } else {
+      return Helper<Ts...>::MakeAudioEncoder(payload_type, format,
+                                             codec_pair_id, field_trials);
+    }
+  }
+};
+
+template <typename... Ts>
+class AudioEncoderFactoryT : public AudioEncoderFactory {
+ public:
+  explicit AudioEncoderFactoryT(const FieldTrialsView* field_trials) {
+    field_trials_ = field_trials;
+  }
+
+  std::vector<AudioCodecSpec> GetSupportedEncoders() override {
+    std::vector<AudioCodecSpec> specs;
+    Helper<Ts...>::AppendSupportedEncoders(&specs);
+    return specs;
+  }
+
+  absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format) override {
+    return Helper<Ts...>::QueryAudioEncoder(format);
+  }
+
+  std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      int payload_type,
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) override {
+    return Helper<Ts...>::MakeAudioEncoder(payload_type, format, codec_pair_id,
+                                           field_trials_);
+  }
+
+  const FieldTrialsView* field_trials_;
+};
+
+}  // namespace audio_encoder_factory_template_impl
+
+// Make an AudioEncoderFactory that can create instances of the given encoders.
+//
+// Each encoder type is given as a template argument to the function; it should
+// be a struct with the following static member functions:
+//
+//   // Converts `audio_format` to a ConfigType instance. Returns an empty
+//   // optional if `audio_format` doesn't correctly specify an encoder of our
+//   // type.
+//   absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
+//
+//   // Appends zero or more AudioCodecSpecs to the list that will be returned
+//   // by AudioEncoderFactory::GetSupportedEncoders().
+//   void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+//
+//   // Returns information about how this format would be encoded. Used to
+//   // implement AudioEncoderFactory::QueryAudioEncoder().
+//   AudioCodecInfo QueryAudioEncoder(const ConfigType& config);
+//
+//   // Creates an AudioEncoder for the specified format. Used to implement
+//   // AudioEncoderFactory::MakeAudioEncoder().
+//   std::unique_ptr<AudioDecoder> MakeAudioEncoder(
+//       const ConfigType& config,
+//       int payload_type,
+//       absl::optional<AudioCodecPairId> codec_pair_id);
+//
+// ConfigType should be a type that encapsulates all the settings needed to
+// create an AudioEncoder. T::Config (where T is the encoder struct) should
+// either be the config type, or an alias for it.
+//
+// Whenever it tries to do something, the new factory will try each of the
+// encoders in the order they were specified in the template argument list,
+// stopping at the first one that claims to be able to do the job.
+//
+// TODO(kwiberg): Point at CreateBuiltinAudioEncoderFactory() for an example of
+// how it is used.
+template <typename... Ts>
+rtc::scoped_refptr<AudioEncoderFactory> CreateAudioEncoderFactory(
+    const FieldTrialsView* field_trials = nullptr) {
+  // There's no technical reason we couldn't allow zero template parameters,
+  // but such a factory couldn't create any encoders, and callers can do this
+  // by mistake by simply forgetting the <> altogether. So we forbid it in
+  // order to prevent caller foot-shooting.
+  static_assert(sizeof...(Ts) >= 1,
+                "Caller must give at least one template parameter");
+
+  return rtc::make_ref_counted<
+      audio_encoder_factory_template_impl::AudioEncoderFactoryT<Ts...>>(
+      field_trials);
+}
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_format.cc b/third_party/libwebrtc/api/audio_codecs/audio_format.cc
new file mode 100644
index 0000000000..2a529a49ee
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_format.cc
@@ -0,0 +1,86 @@
+/*
+ *  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 "api/audio_codecs/audio_format.h"
+
+#include <utility>
+
+#include "absl/strings/match.h"
+
+namespace webrtc {
+
+SdpAudioFormat::SdpAudioFormat(const SdpAudioFormat&) = default;
+SdpAudioFormat::SdpAudioFormat(SdpAudioFormat&&) = default;
+
+SdpAudioFormat::SdpAudioFormat(absl::string_view name,
+                               int clockrate_hz,
+                               size_t num_channels)
+    : name(name), clockrate_hz(clockrate_hz), num_channels(num_channels) {}
+
+SdpAudioFormat::SdpAudioFormat(absl::string_view name,
+                               int clockrate_hz,
+                               size_t num_channels,
+                               const Parameters& param)
+    : name(name),
+      clockrate_hz(clockrate_hz),
+      num_channels(num_channels),
+      parameters(param) {}
+
+SdpAudioFormat::SdpAudioFormat(absl::string_view name,
+                               int clockrate_hz,
+                               size_t num_channels,
+                               Parameters&& param)
+    : name(name),
+      clockrate_hz(clockrate_hz),
+      num_channels(num_channels),
+      parameters(std::move(param)) {}
+
+bool SdpAudioFormat::Matches(const SdpAudioFormat& o) const {
+  return absl::EqualsIgnoreCase(name, o.name) &&
+         clockrate_hz == o.clockrate_hz && num_channels == o.num_channels;
+}
+
+SdpAudioFormat::~SdpAudioFormat() = default;
+SdpAudioFormat& SdpAudioFormat::operator=(const SdpAudioFormat&) = default;
+SdpAudioFormat& SdpAudioFormat::operator=(SdpAudioFormat&&) = default;
+
+bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b) {
+  return absl::EqualsIgnoreCase(a.name, b.name) &&
+         a.clockrate_hz == b.clockrate_hz && a.num_channels == b.num_channels &&
+         a.parameters == b.parameters;
+}
+
+AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
+                               size_t num_channels,
+                               int bitrate_bps)
+    : AudioCodecInfo(sample_rate_hz,
+                     num_channels,
+                     bitrate_bps,
+                     bitrate_bps,
+                     bitrate_bps) {}
+
+AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
+                               size_t num_channels,
+                               int default_bitrate_bps,
+                               int min_bitrate_bps,
+                               int max_bitrate_bps)
+    : sample_rate_hz(sample_rate_hz),
+      num_channels(num_channels),
+      default_bitrate_bps(default_bitrate_bps),
+      min_bitrate_bps(min_bitrate_bps),
+      max_bitrate_bps(max_bitrate_bps) {
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  RTC_DCHECK_GT(num_channels, 0);
+  RTC_DCHECK_GE(min_bitrate_bps, 0);
+  RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps);
+  RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/audio_format.h b/third_party/libwebrtc/api/audio_codecs/audio_format.h
new file mode 100644
index 0000000000..0cf67799b8
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/audio_format.h
@@ -0,0 +1,133 @@
+/*
+ *  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 API_AUDIO_CODECS_AUDIO_FORMAT_H_
+#define API_AUDIO_CODECS_AUDIO_FORMAT_H_
+
+#include <stddef.h>
+
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// SDP specification for a single audio codec.
+struct RTC_EXPORT SdpAudioFormat {
+  using Parameters = std::map<std::string, std::string>;
+
+  SdpAudioFormat(const SdpAudioFormat&);
+  SdpAudioFormat(SdpAudioFormat&&);
+  SdpAudioFormat(absl::string_view name, int clockrate_hz, size_t num_channels);
+  SdpAudioFormat(absl::string_view name,
+                 int clockrate_hz,
+                 size_t num_channels,
+                 const Parameters& param);
+  SdpAudioFormat(absl::string_view name,
+                 int clockrate_hz,
+                 size_t num_channels,
+                 Parameters&& param);
+  ~SdpAudioFormat();
+
+  // Returns true if this format is compatible with `o`. In SDP terminology:
+  // would it represent the same codec between an offer and an answer? As
+  // opposed to operator==, this method disregards codec parameters.
+  bool Matches(const SdpAudioFormat& o) const;
+
+  SdpAudioFormat& operator=(const SdpAudioFormat&);
+  SdpAudioFormat& operator=(SdpAudioFormat&&);
+
+  friend bool operator==(const SdpAudioFormat& a, const SdpAudioFormat& b);
+  friend bool operator!=(const SdpAudioFormat& a, const SdpAudioFormat& b) {
+    return !(a == b);
+  }
+
+  std::string name;
+  int clockrate_hz;
+  size_t num_channels;
+  Parameters parameters;
+};
+
+// Information about how an audio format is treated by the codec implementation.
+// Contains basic information, such as sample rate and number of channels, which
+// isn't uniformly presented by SDP. Also contains flags indicating support for
+// integrating with other parts of WebRTC, like external VAD and comfort noise
+// level calculation.
+//
+// To avoid API breakage, and make the code clearer, AudioCodecInfo should not
+// be directly initializable with any flags indicating optional support. If it
+// were, these initializers would break any time a new flag was added. It's also
+// more difficult to understand:
+//   AudioCodecInfo info{16000, 1, 32000, true, false, false, true, true};
+// than
+//   AudioCodecInfo info(16000, 1, 32000);
+//   info.allow_comfort_noise = true;
+//   info.future_flag_b = true;
+//   info.future_flag_c = true;
+struct AudioCodecInfo {
+  AudioCodecInfo(int sample_rate_hz, size_t num_channels, int bitrate_bps);
+  AudioCodecInfo(int sample_rate_hz,
+                 size_t num_channels,
+                 int default_bitrate_bps,
+                 int min_bitrate_bps,
+                 int max_bitrate_bps);
+  AudioCodecInfo(const AudioCodecInfo& b) = default;
+  ~AudioCodecInfo() = default;
+
+  bool operator==(const AudioCodecInfo& b) const {
+    return sample_rate_hz == b.sample_rate_hz &&
+           num_channels == b.num_channels &&
+           default_bitrate_bps == b.default_bitrate_bps &&
+           min_bitrate_bps == b.min_bitrate_bps &&
+           max_bitrate_bps == b.max_bitrate_bps &&
+           allow_comfort_noise == b.allow_comfort_noise &&
+           supports_network_adaption == b.supports_network_adaption;
+  }
+
+  bool operator!=(const AudioCodecInfo& b) const { return !(*this == b); }
+
+  bool HasFixedBitrate() const {
+    RTC_DCHECK_GE(min_bitrate_bps, 0);
+    RTC_DCHECK_LE(min_bitrate_bps, default_bitrate_bps);
+    RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
+    return min_bitrate_bps == max_bitrate_bps;
+  }
+
+  int sample_rate_hz;
+  size_t num_channels;
+  int default_bitrate_bps;
+  int min_bitrate_bps;
+  int max_bitrate_bps;
+
+  bool allow_comfort_noise = true;  // This codec can be used with an external
+                                    // comfort noise generator.
+  bool supports_network_adaption = false;  // This codec can adapt to varying
+                                           // network conditions.
+};
+
+// AudioCodecSpec ties an audio format to specific information about the codec
+// and its implementation.
+struct AudioCodecSpec {
+  bool operator==(const AudioCodecSpec& b) const {
+    return format == b.format && info == b.info;
+  }
+
+  bool operator!=(const AudioCodecSpec& b) const { return !(*this == b); }
+
+  SdpAudioFormat format;
+  AudioCodecInfo info;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_AUDIO_FORMAT_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.cc b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.cc
new file mode 100644
index 0000000000..881113d985
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.cc
@@ -0,0 +1,68 @@
+/*
+ *  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 "api/audio_codecs/builtin_audio_decoder_factory.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/L16/audio_decoder_L16.h"
+#include "api/audio_codecs/audio_decoder_factory_template.h"
+#include "api/audio_codecs/g711/audio_decoder_g711.h"
+#include "api/audio_codecs/g722/audio_decoder_g722.h"
+#if WEBRTC_USE_BUILTIN_ILBC
+#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"  // nogncheck
+#endif
+#if WEBRTC_USE_BUILTIN_OPUS
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_decoder_opus.h"  // nogncheck
+#endif
+
+namespace webrtc {
+
+namespace {
+
+// Modify an audio decoder to not advertise support for anything.
+template <typename T>
+struct NotAdvertised {
+  using Config = typename T::Config;
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    return T::SdpToConfig(audio_format);
+  }
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
+    // Don't advertise support for anything.
+  }
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config& config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
+    return T::MakeAudioDecoder(config, codec_pair_id);
+  }
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
+  return CreateAudioDecoderFactory<
+
+#if WEBRTC_USE_BUILTIN_OPUS
+      AudioDecoderOpus, NotAdvertised<AudioDecoderMultiChannelOpus>,
+#endif
+
+      AudioDecoderG722,
+
+#if WEBRTC_USE_BUILTIN_ILBC
+      AudioDecoderIlbc,
+#endif
+
+      AudioDecoderG711, NotAdvertised<AudioDecoderL16>>();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.h b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.h
new file mode 100644
index 0000000000..72e1e3d96e
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory.h
@@ -0,0 +1,28 @@
+/*
+ *  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 API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+#define API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create the built-in types of audio decoders.
+// Note: This will link with all the code implementing those codecs, so if you
+// only need a subset of the codecs, consider using
+// CreateAudioDecoderFactory<...codecs listed here...>() or
+// CreateOpusAudioDecoderFactory() instead.
+rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory();
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory_gn/moz.build
new file mode 100644
index 0000000000..366307ea13
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_decoder_factory_gn/moz.build
@@ -0,0 +1,234 @@
+# 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"
+DEFINES["WEBRTC_USE_BUILTIN_ILBC"] = "1"
+DEFINES["WEBRTC_USE_BUILTIN_OPUS"] = "1"
+
+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/api/audio_codecs/builtin_audio_decoder_factory.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_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_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_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["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["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("builtin_audio_decoder_factory_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.cc b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.cc
new file mode 100644
index 0000000000..4546a2eaee
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.cc
@@ -0,0 +1,74 @@
+/*
+ *  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 "api/audio_codecs/builtin_audio_encoder_factory.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/L16/audio_encoder_L16.h"
+#include "api/audio_codecs/audio_encoder_factory_template.h"
+#include "api/audio_codecs/g711/audio_encoder_g711.h"
+#include "api/audio_codecs/g722/audio_encoder_g722.h"
+#if WEBRTC_USE_BUILTIN_ILBC
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"  // nogncheck
+#endif
+#if WEBRTC_USE_BUILTIN_OPUS
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"  // nogncheck
+#endif
+
+namespace webrtc {
+
+namespace {
+
+// Modify an audio encoder to not advertise support for anything.
+template <typename T>
+struct NotAdvertised {
+  using Config = typename T::Config;
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    return T::SdpToConfig(audio_format);
+  }
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
+    // Don't advertise support for anything.
+  }
+  static AudioCodecInfo QueryAudioEncoder(const Config& config) {
+    return T::QueryAudioEncoder(config);
+  }
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr) {
+    return T::MakeAudioEncoder(config, payload_type, codec_pair_id,
+                               field_trials);
+  }
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory() {
+  return CreateAudioEncoderFactory<
+
+#if WEBRTC_USE_BUILTIN_OPUS
+      AudioEncoderOpus, NotAdvertised<AudioEncoderMultiChannelOpus>,
+#endif
+
+      AudioEncoderG722,
+
+#if WEBRTC_USE_BUILTIN_ILBC
+      AudioEncoderIlbc,
+#endif
+
+      AudioEncoderG711, NotAdvertised<AudioEncoderL16>>();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.h b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.h
new file mode 100644
index 0000000000..f833de10f1
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory.h
@@ -0,0 +1,28 @@
+/*
+ *  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 API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
+#define API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create the built-in types of audio encoders.
+// Note: This will link with all the code implementing those codecs, so if you
+// only need a subset of the codecs, consider using
+// CreateAudioEncoderFactory<...codecs listed here...>() or
+// CreateOpusAudioEncoderFactory() instead.
+rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory();
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory_gn/moz.build
new file mode 100644
index 0000000000..db0e3fbe00
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/builtin_audio_encoder_factory_gn/moz.build
@@ -0,0 +1,234 @@
+# 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"
+DEFINES["WEBRTC_USE_BUILTIN_ILBC"] = "1"
+DEFINES["WEBRTC_USE_BUILTIN_OPUS"] = "1"
+
+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/api/audio_codecs/builtin_audio_encoder_factory.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_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_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_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["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["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("builtin_audio_encoder_factory_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/g711/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/g711/BUILD.gn
new file mode 100644
index 0000000000..b2ff324f12
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g711/BUILD.gn
@@ -0,0 +1,55 @@
+# 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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio_encoder_g711") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_g711.cc",
+    "audio_encoder_g711.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g711",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_g711") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_g711.cc",
+    "audio_decoder_g711.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g711",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.cc b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.cc
new file mode 100644
index 0000000000..838f7e9624
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.cc
@@ -0,0 +1,67 @@
+/*
+ *  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 "api/audio_codecs/g711/audio_decoder_g711.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
+    const SdpAudioFormat& format) {
+  const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
+  const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
+  if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
+      (is_pcmu || is_pcma)) {
+    Config config;
+    config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
+    config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
+    return config;
+  } else {
+    return absl::nullopt;
+  }
+}
+
+void AudioDecoderG711::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  for (const char* type : {"PCMU", "PCMA"}) {
+    specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
+  }
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
+    const Config& config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  switch (config.type) {
+    case Config::Type::kPcmU:
+      return std::make_unique<AudioDecoderPcmU>(config.num_channels);
+    case Config::Type::kPcmA:
+      return std::make_unique<AudioDecoderPcmA>(config.num_channels);
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return nullptr;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.h b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.h
new file mode 100644
index 0000000000..0f7a98d345
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.h
@@ -0,0 +1,49 @@
+/*
+ *  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 API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
+#define API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// G711 decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderG711 {
+  struct Config {
+    enum class Type { kPcmU, kPcmA };
+    bool IsOk() const {
+      return (type == Type::kPcmU || type == Type::kPcmA) &&
+             num_channels >= 1 &&
+             num_channels <= AudioDecoder::kMaxNumberOfChannels;
+    }
+    Type type;
+    int num_channels;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config& config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G711_AUDIO_DECODER_G711_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711_gn/moz.build
new file mode 100644
index 0000000000..4782d01dd1
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711_gn/moz.build
@@ -0,0 +1,225 @@
+# 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/api/audio_codecs/g711/audio_decoder_g711.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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_decoder_g711_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.cc b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.cc
new file mode 100644
index 0000000000..1dca3b80d3
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.cc
@@ -0,0 +1,95 @@
+/*
+ *  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 "api/audio_codecs/g711/audio_encoder_g711.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
+    const SdpAudioFormat& format) {
+  const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
+  const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
+  if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
+      (is_pcmu || is_pcma)) {
+    Config config;
+    config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
+    config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
+    config.frame_size_ms = 20;
+    auto ptime_iter = format.parameters.find("ptime");
+    if (ptime_iter != format.parameters.end()) {
+      const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+      if (ptime && *ptime > 0) {
+        config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
+      }
+    }
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
+    return config;
+  } else {
+    return absl::nullopt;
+  }
+}
+
+void AudioEncoderG711::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  for (const char* type : {"PCMU", "PCMA"}) {
+    specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
+  }
+}
+
+AudioCodecInfo AudioEncoderG711::QueryAudioEncoder(const Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {8000, rtc::dchecked_cast<size_t>(config.num_channels),
+          64000 * config.num_channels};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
+    const Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  switch (config.type) {
+    case Config::Type::kPcmU: {
+      AudioEncoderPcmU::Config impl_config;
+      impl_config.num_channels = config.num_channels;
+      impl_config.frame_size_ms = config.frame_size_ms;
+      impl_config.payload_type = payload_type;
+      return std::make_unique<AudioEncoderPcmU>(impl_config);
+    }
+    case Config::Type::kPcmA: {
+      AudioEncoderPcmA::Config impl_config;
+      impl_config.num_channels = config.num_channels;
+      impl_config.frame_size_ms = config.frame_size_ms;
+      impl_config.payload_type = payload_type;
+      return std::make_unique<AudioEncoderPcmA>(impl_config);
+    }
+    default: {
+      RTC_DCHECK_NOTREACHED();
+      return nullptr;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.h b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.h
new file mode 100644
index 0000000000..4b3eb845e0
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.h
@@ -0,0 +1,54 @@
+/*
+ *  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 API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
+#define API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// G711 encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderG711 {
+  struct Config {
+    enum class Type { kPcmU, kPcmA };
+    bool IsOk() const {
+      return (type == Type::kPcmU || type == Type::kPcmA) &&
+             frame_size_ms > 0 && frame_size_ms % 10 == 0 &&
+             num_channels >= 1 &&
+             num_channels <= AudioEncoder::kMaxNumberOfChannels;
+    }
+    Type type = Type::kPcmU;
+    int num_channels = 1;
+    int frame_size_ms = 20;
+  };
+  static absl::optional<AudioEncoderG711::Config> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G711_AUDIO_ENCODER_G711_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711_gn/moz.build
new file mode 100644
index 0000000000..c972978c13
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711_gn/moz.build
@@ -0,0 +1,225 @@
+# 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/api/audio_codecs/g711/audio_encoder_g711.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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_encoder_g711_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/g722/BUILD.gn
new file mode 100644
index 0000000000..af13ac3de3
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g722/BUILD.gn
@@ -0,0 +1,62 @@
+# 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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_source_set("audio_encoder_g722_config") {
+  visibility = [ "*" ]
+  sources = [ "audio_encoder_g722_config.h" ]
+  deps = [ "..:audio_codecs_api" ]
+}
+
+rtc_library("audio_encoder_g722") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_g722.cc",
+    "audio_encoder_g722.h",
+  ]
+  deps = [
+    ":audio_encoder_g722_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g722",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_g722") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_g722.cc",
+    "audio_decoder_g722.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:g722",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.cc b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.cc
new file mode 100644
index 0000000000..ed7163471a
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.cc
@@ -0,0 +1,56 @@
+/*
+ *  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 "api/audio_codecs/g722/audio_decoder_g722.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (absl::EqualsIgnoreCase(format.name, "G722") &&
+      format.clockrate_hz == 8000 &&
+      (format.num_channels == 1 || format.num_channels == 2)) {
+    return Config{rtc::dchecked_cast<int>(format.num_channels)};
+  }
+  return absl::nullopt;
+}
+
+void AudioDecoderG722::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  specs->push_back({{"G722", 8000, 1}, {16000, 1, 64000}});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder(
+    Config config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  switch (config.num_channels) {
+    case 1:
+      return std::make_unique<AudioDecoderG722Impl>();
+    case 2:
+      return std::make_unique<AudioDecoderG722StereoImpl>();
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return nullptr;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.h b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.h
new file mode 100644
index 0000000000..6f7b253039
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.h
@@ -0,0 +1,43 @@
+/*
+ *  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 API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
+#define API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// G722 decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderG722 {
+  struct Config {
+    bool IsOk() const { return num_channels == 1 || num_channels == 2; }
+    int num_channels;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      Config config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G722_AUDIO_DECODER_G722_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722_gn/moz.build
new file mode 100644
index 0000000000..77003c77a9
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722_gn/moz.build
@@ -0,0 +1,225 @@
+# 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/api/audio_codecs/g722/audio_decoder_g722.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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_decoder_g722_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc
new file mode 100644
index 0000000000..56a6c4da6a
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc
@@ -0,0 +1,74 @@
+/*
+ *  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 "api/audio_codecs/g722/audio_encoder_g722.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name, "g722") ||
+      format.clockrate_hz != 8000) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderG722Config config;
+  config.num_channels = rtc::checked_cast<int>(format.num_channels);
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      const int whole_packets = *ptime / 10;
+      config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
+    }
+  }
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+void AudioEncoderG722::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"G722", 8000, 1};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
+    const AudioEncoderG722Config& config) {
+  RTC_DCHECK(config.IsOk());
+  return {16000, rtc::dchecked_cast<size_t>(config.num_channels),
+          64000 * config.num_channels};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
+    const AudioEncoderG722Config& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderG722Impl>(config, payload_type);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.h b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.h
new file mode 100644
index 0000000000..78ceddd1e9
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.h
@@ -0,0 +1,44 @@
+/*
+ *  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 API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
+#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// G722 encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderG722 {
+  using Config = AudioEncoderG722Config;
+  static absl::optional<AudioEncoderG722Config> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderG722Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderG722Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config.h b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config.h
new file mode 100644
index 0000000000..f3f3a9f016
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config.h
@@ -0,0 +1,29 @@
+/*
+ *  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 API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
+#define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
+
+#include "api/audio_codecs/audio_encoder.h"
+
+namespace webrtc {
+
+struct AudioEncoderG722Config {
+  bool IsOk() const {
+    return frame_size_ms > 0 && frame_size_ms % 10 == 0 && num_channels >= 1 &&
+           num_channels <= AudioEncoder::kMaxNumberOfChannels;
+  }
+  int frame_size_ms = 20;
+  int num_channels = 1;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config_gn/moz.build
new file mode 100644
index 0000000000..41e1e248c5
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_config_gn/moz.build
@@ -0,0 +1,209 @@
+# 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"
+]
+
+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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_encoder_g722_config_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_gn/moz.build
new file mode 100644
index 0000000000..c3beba6cdb
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722_gn/moz.build
@@ -0,0 +1,225 @@
+# 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/api/audio_codecs/g722/audio_encoder_g722.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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_encoder_g722_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/ilbc/BUILD.gn
new file mode 100644
index 0000000000..22cf48220f
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/BUILD.gn
@@ -0,0 +1,58 @@
+# 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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_source_set("audio_encoder_ilbc_config") {
+  visibility = [ "*" ]
+  sources = [ "audio_encoder_ilbc_config.h" ]
+}
+
+rtc_library("audio_encoder_ilbc") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_encoder_ilbc.cc",
+    "audio_encoder_ilbc.h",
+  ]
+  deps = [
+    ":audio_encoder_ilbc_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:ilbc",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base:stringutils",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_ilbc") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_ilbc.cc",
+    "audio_decoder_ilbc.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:ilbc",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
new file mode 100644
index 0000000000..c58316903a
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
@@ -0,0 +1,42 @@
+/*
+ *  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 "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (absl::EqualsIgnoreCase(format.name, "ILBC") &&
+      format.clockrate_hz == 8000 && format.num_channels == 1) {
+    return Config();
+  }
+  return absl::nullopt;
+}
+
+void AudioDecoderIlbc::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  specs->push_back({{"ILBC", 8000, 1}, {8000, 1, 13300}});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderIlbc::MakeAudioDecoder(
+    Config config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  return std::make_unique<AudioDecoderIlbcImpl>();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
new file mode 100644
index 0000000000..60566c88df
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
@@ -0,0 +1,39 @@
+/*
+ *  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 API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#define API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+
+namespace webrtc {
+
+// ILBC decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct AudioDecoderIlbc {
+  struct Config {};  // Empty---no config values needed!
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      Config config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc_gn/moz.build
new file mode 100644
index 0000000000..53e9d1a4a7
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc_gn/moz.build
@@ -0,0 +1,232 @@
+# 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/api/audio_codecs/ilbc/audio_decoder_ilbc.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_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_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_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["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["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_decoder_ilbc_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
new file mode 100644
index 0000000000..b497948491
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
@@ -0,0 +1,88 @@
+/*
+ *  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 "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+namespace {
+int GetIlbcBitrate(int ptime) {
+  switch (ptime) {
+    case 20:
+    case 40:
+      // 38 bytes per frame of 20 ms => 15200 bits/s.
+      return 15200;
+    case 30:
+    case 60:
+      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
+      return 13333;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+}  // namespace
+
+absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") ||
+      format.clockrate_hz != 8000 || format.num_channels != 1) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderIlbcConfig config;
+  auto ptime_iter = format.parameters.find("ptime");
+  if (ptime_iter != format.parameters.end()) {
+    auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
+    if (ptime && *ptime > 0) {
+      const int whole_packets = *ptime / 10;
+      config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 20, 60);
+    }
+  }
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+void AudioEncoderIlbc::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"ILBC", 8000, 1};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder(
+    const AudioEncoderIlbcConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return {8000, 1, GetIlbcBitrate(config.frame_size_ms)};
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
+    const AudioEncoderIlbcConfig& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderIlbcImpl>(config, payload_type);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
new file mode 100644
index 0000000000..a5306841ce
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
@@ -0,0 +1,43 @@
+/*
+ *  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 API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
+#include "api/field_trials_view.h"
+
+namespace webrtc {
+
+// ILBC encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct AudioEncoderIlbc {
+  using Config = AudioEncoderIlbcConfig;
+  static absl::optional<AudioEncoderIlbcConfig> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderIlbcConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderIlbcConfig& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h
new file mode 100644
index 0000000000..4d82f9901c
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h
@@ -0,0 +1,28 @@
+/*
+ *  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 API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
+#define API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
+
+namespace webrtc {
+
+struct AudioEncoderIlbcConfig {
+  bool IsOk() const {
+    return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
+            frame_size_ms == 60);
+  }
+  int frame_size_ms = 30;  // Valid values are 20, 30, 40, and 60 ms.
+  // Note that frame size 40 ms produces encodings with two 20 ms frames in
+  // them, and frame size 60 ms consists of two 30 ms frames.
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_ILBC_AUDIO_ENCODER_ILBC_CONFIG_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config_gn/moz.build
new file mode 100644
index 0000000000..75737b8f19
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config_gn/moz.build
@@ -0,0 +1,201 @@
+# 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"
+]
+
+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_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_encoder_ilbc_config_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_gn/moz.build
new file mode 100644
index 0000000000..bddfe42193
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_gn/moz.build
@@ -0,0 +1,232 @@
+# 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/api/audio_codecs/ilbc/audio_encoder_ilbc.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_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_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_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["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["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_encoder_ilbc_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn
new file mode 100644
index 0000000000..eb90a0b9ac
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn
@@ -0,0 +1,110 @@
+# 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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+rtc_library("audio_encoder_opus_config") {
+  visibility = [ "*" ]
+  sources = [
+    "audio_encoder_multi_channel_opus_config.cc",
+    "audio_encoder_multi_channel_opus_config.h",
+    "audio_encoder_opus_config.cc",
+    "audio_encoder_opus_config.h",
+  ]
+  deps = [ "../../../rtc_base/system:rtc_export" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  defines = []
+  if (rtc_opus_variable_complexity) {
+    defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
+  } else {
+    defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
+  }
+}
+
+rtc_source_set("audio_decoder_opus_config") {
+  visibility = [ "*" ]
+  sources = [ "audio_decoder_multi_channel_opus_config.h" ]
+  deps = [ "..:audio_codecs_api" ]
+}
+
+rtc_library("audio_encoder_opus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  public = [ "audio_encoder_opus.h" ]
+  sources = [ "audio_encoder_opus.cc" ]
+  deps = [
+    ":audio_encoder_opus_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_opus",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_decoder_opus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_opus.cc",
+    "audio_decoder_opus.h",
+  ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_opus",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("audio_encoder_multiopus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  public = [ "audio_encoder_multi_channel_opus.h" ]
+  sources = [ "audio_encoder_multi_channel_opus.cc" ]
+  deps = [
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_multiopus",
+    "../../../rtc_base/system:rtc_export",
+    "../opus:audio_encoder_opus_config",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("audio_decoder_multiopus") {
+  visibility = [ "*" ]
+  poisonous = [ "audio_codecs" ]
+  sources = [
+    "audio_decoder_multi_channel_opus.cc",
+    "audio_decoder_multi_channel_opus.h",
+  ]
+  deps = [
+    ":audio_decoder_opus_config",
+    "..:audio_codecs_api",
+    "../../../api:field_trials_view",
+    "../../../modules/audio_coding:webrtc_multiopus",
+    "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc
new file mode 100644
index 0000000000..0fb4e05511
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc
@@ -0,0 +1,71 @@
+/*
+ *  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 "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h"
+
+namespace webrtc {
+
+absl::optional<AudioDecoderMultiChannelOpusConfig>
+AudioDecoderMultiChannelOpus::SdpToConfig(const SdpAudioFormat& format) {
+  return AudioDecoderMultiChannelOpusImpl::SdpToConfig(format);
+}
+
+void AudioDecoderMultiChannelOpus::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  // To get full utilization of the surround support of the Opus lib, we can
+  // mark which channel is the low frequency effects (LFE). But that is not done
+  // ATM.
+  {
+    AudioCodecInfo surround_5_1_opus_info{48000, 6,
+                                          /* default_bitrate_bps= */ 128000};
+    surround_5_1_opus_info.allow_comfort_noise = false;
+    surround_5_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                6,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,4,1,2,3,5"},
+                                 {"num_streams", "4"},
+                                 {"coupled_streams", "2"}}});
+    specs->push_back({std::move(opus_format), surround_5_1_opus_info});
+  }
+  {
+    AudioCodecInfo surround_7_1_opus_info{48000, 8,
+                                          /* default_bitrate_bps= */ 200000};
+    surround_7_1_opus_info.allow_comfort_noise = false;
+    surround_7_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                8,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,6,1,2,3,4,5,7"},
+                                 {"num_streams", "5"},
+                                 {"coupled_streams", "3"}}});
+    specs->push_back({std::move(opus_format), surround_7_1_opus_info});
+  }
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderMultiChannelOpus::MakeAudioDecoder(
+    AudioDecoderMultiChannelOpusConfig config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  return AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(config);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h
new file mode 100644
index 0000000000..eafd6c6939
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.h
@@ -0,0 +1,42 @@
+/*
+ *  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 API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderMultiChannelOpus {
+  using Config = AudioDecoderMultiChannelOpusConfig;
+  static absl::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      AudioDecoderMultiChannelOpusConfig config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h
new file mode 100644
index 0000000000..f97c5c3193
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h
@@ -0,0 +1,66 @@
+/*
+ *  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 API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+
+namespace webrtc {
+struct AudioDecoderMultiChannelOpusConfig {
+  // The number of channels that the decoder will output.
+  int num_channels;
+
+  // Number of mono or stereo encoded Opus streams.
+  int num_streams;
+
+  // Number of channel pairs coupled together, see RFC 7845 section
+  // 5.1.1. Has to be less than the number of streams.
+  int coupled_streams;
+
+  // Channel mapping table, defines the mapping from encoded streams to output
+  // channels. See RFC 7845 section 5.1.1.
+  std::vector<unsigned char> channel_mapping;
+
+  bool IsOk() const {
+    if (num_channels < 1 || num_channels > AudioDecoder::kMaxNumberOfChannels ||
+        num_streams < 0 || coupled_streams < 0) {
+      return false;
+    }
+    if (num_streams < coupled_streams) {
+      return false;
+    }
+    if (channel_mapping.size() != static_cast<size_t>(num_channels)) {
+      return false;
+    }
+
+    // Every mono stream codes one channel, every coupled stream codes two. This
+    // is the total coded channel count:
+    const int max_coded_channel = num_streams + coupled_streams;
+    for (const auto& x : channel_mapping) {
+      // Coded channels >= max_coded_channel don't exist. Except for 255, which
+      // tells Opus to put silence in output channel x.
+      if (x >= max_coded_channel && x != 255) {
+        return false;
+      }
+    }
+
+    if (num_channels > 255 || max_coded_channel >= 255) {
+      return false;
+    }
+    return true;
+  }
+};
+
+}  // namespace webrtc
+
+#endif  //  API_AUDIO_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_CONFIG_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multiopus_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multiopus_gn/moz.build
new file mode 100644
index 0000000000..2b2bc6d9a7
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multiopus_gn/moz.build
@@ -0,0 +1,226 @@
+# 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",
+    "/media/libopus/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_decoder_multiopus_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc
new file mode 100644
index 0000000000..efc9a73546
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.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 "api/audio_codecs/opus/audio_decoder_opus.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+
+namespace webrtc {
+
+bool AudioDecoderOpus::Config::IsOk() const {
+  if (sample_rate_hz != 16000 && sample_rate_hz != 48000) {
+    // Unsupported sample rate. (libopus supports a few other rates as
+    // well; we can add support for them when needed.)
+    return false;
+  }
+  if (num_channels != 1 && num_channels != 2) {
+    return false;
+  }
+  return true;
+}
+
+absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
+    const SdpAudioFormat& format) {
+  const auto num_channels = [&]() -> absl::optional<int> {
+    auto stereo = format.parameters.find("stereo");
+    if (stereo != format.parameters.end()) {
+      if (stereo->second == "0") {
+        return 1;
+      } else if (stereo->second == "1") {
+        return 2;
+      } else {
+        return absl::nullopt;  // Bad stereo parameter.
+      }
+    }
+    return 1;  // Default to mono.
+  }();
+  if (absl::EqualsIgnoreCase(format.name, "opus") &&
+      format.clockrate_hz == 48000 && format.num_channels == 2 &&
+      num_channels) {
+    Config config;
+    config.num_channels = *num_channels;
+    if (!config.IsOk()) {
+      RTC_DCHECK_NOTREACHED();
+      return absl::nullopt;
+    }
+    return config;
+  } else {
+    return absl::nullopt;
+  }
+}
+
+void AudioDecoderOpus::AppendSupportedDecoders(
+    std::vector<AudioCodecSpec>* specs) {
+  AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
+  opus_info.allow_comfort_noise = false;
+  opus_info.supports_network_adaption = true;
+  SdpAudioFormat opus_format(
+      {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}});
+  specs->push_back({std::move(opus_format), opus_info});
+}
+
+std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
+    Config config,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioDecoderOpusImpl>(config.num_channels,
+                                                config.sample_rate_hz);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.h
new file mode 100644
index 0000000000..138c0377df
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.h
@@ -0,0 +1,44 @@
+/*
+ *  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 API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus decoder API for use as a template parameter to
+// CreateAudioDecoderFactory<...>().
+struct RTC_EXPORT AudioDecoderOpus {
+  struct Config {
+    bool IsOk() const;  // Checks if the values are currently OK.
+    int sample_rate_hz = 48000;
+    int num_channels = 1;
+  };
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      Config config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_config_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_config_gn/moz.build
new file mode 100644
index 0000000000..e2c470d5ee
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_config_gn/moz.build
@@ -0,0 +1,209 @@
+# 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"
+]
+
+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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_decoder_opus_config_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_gn/moz.build
new file mode 100644
index 0000000000..58e6355a55
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus_gn/moz.build
@@ -0,0 +1,233 @@
+# 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",
+    "/media/libopus/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.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_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_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_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["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["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_decoder_opus_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc
new file mode 100644
index 0000000000..14f480b1ec
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.cc
@@ -0,0 +1,75 @@
+/*
+ *  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 "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderMultiChannelOpusConfig>
+AudioEncoderMultiChannelOpus::SdpToConfig(const SdpAudioFormat& format) {
+  return AudioEncoderMultiChannelOpusImpl::SdpToConfig(format);
+}
+
+void AudioEncoderMultiChannelOpus::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  // To get full utilization of the surround support of the Opus lib, we can
+  // mark which channel is the low frequency effects (LFE). But that is not done
+  // ATM.
+  {
+    AudioCodecInfo surround_5_1_opus_info{48000, 6,
+                                          /* default_bitrate_bps= */ 128000};
+    surround_5_1_opus_info.allow_comfort_noise = false;
+    surround_5_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                6,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,4,1,2,3,5"},
+                                 {"num_streams", "4"},
+                                 {"coupled_streams", "2"}}});
+    specs->push_back({std::move(opus_format), surround_5_1_opus_info});
+  }
+  {
+    AudioCodecInfo surround_7_1_opus_info{48000, 8,
+                                          /* default_bitrate_bps= */ 200000};
+    surround_7_1_opus_info.allow_comfort_noise = false;
+    surround_7_1_opus_info.supports_network_adaption = false;
+    SdpAudioFormat opus_format({"multiopus",
+                                48000,
+                                8,
+                                {{"minptime", "10"},
+                                 {"useinbandfec", "1"},
+                                 {"channel_mapping", "0,6,1,2,3,4,5,7"},
+                                 {"num_streams", "5"},
+                                 {"coupled_streams", "3"}}});
+    specs->push_back({std::move(opus_format), surround_7_1_opus_info});
+  }
+}
+
+AudioCodecInfo AudioEncoderMultiChannelOpus::QueryAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config) {
+  return AudioEncoderMultiChannelOpusImpl::QueryAudioEncoder(config);
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderMultiChannelOpus::MakeAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  return AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(config,
+                                                            payload_type);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h
new file mode 100644
index 0000000000..c1c4db3577
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.h
@@ -0,0 +1,43 @@
+/*
+ *  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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderMultiChannelOpus {
+  using Config = AudioEncoderMultiChannelOpusConfig;
+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const Config& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc
new file mode 100644
index 0000000000..0052c429b2
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.cc
@@ -0,0 +1,106 @@
+/*
+ *  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 "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kDefaultComplexity = 9;
+}  // namespace
+
+AudioEncoderMultiChannelOpusConfig::AudioEncoderMultiChannelOpusConfig()
+    : frame_size_ms(kDefaultFrameSizeMs),
+      num_channels(1),
+      application(ApplicationMode::kVoip),
+      bitrate_bps(32000),
+      fec_enabled(false),
+      cbr_enabled(false),
+      dtx_enabled(false),
+      max_playback_rate_hz(48000),
+      complexity(kDefaultComplexity),
+      num_streams(-1),
+      coupled_streams(-1) {}
+AudioEncoderMultiChannelOpusConfig::AudioEncoderMultiChannelOpusConfig(
+    const AudioEncoderMultiChannelOpusConfig&) = default;
+AudioEncoderMultiChannelOpusConfig::~AudioEncoderMultiChannelOpusConfig() =
+    default;
+AudioEncoderMultiChannelOpusConfig& AudioEncoderMultiChannelOpusConfig::
+operator=(const AudioEncoderMultiChannelOpusConfig&) = default;
+
+bool AudioEncoderMultiChannelOpusConfig::IsOk() const {
+  if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
+    return false;
+  if (num_channels >= 255) {
+    return false;
+  }
+  if (bitrate_bps < kMinBitrateBps || bitrate_bps > kMaxBitrateBps)
+    return false;
+  if (complexity < 0 || complexity > 10)
+    return false;
+
+  // Check the lengths:
+  if (num_streams < 0 || coupled_streams < 0) {
+    return false;
+  }
+  if (num_streams < coupled_streams) {
+    return false;
+  }
+  if (channel_mapping.size() != static_cast<size_t>(num_channels)) {
+    return false;
+  }
+
+  // Every mono stream codes one channel, every coupled stream codes two. This
+  // is the total coded channel count:
+  const int max_coded_channel = num_streams + coupled_streams;
+  for (const auto& x : channel_mapping) {
+    // Coded channels >= max_coded_channel don't exist. Except for 255, which
+    // tells Opus to ignore input channel x.
+    if (x >= max_coded_channel && x != 255) {
+      return false;
+    }
+  }
+
+  // Inverse mapping.
+  constexpr int kNotSet = -1;
+  std::vector<int> coded_channels_to_input_channels(max_coded_channel, kNotSet);
+  for (size_t i = 0; i < num_channels; ++i) {
+    if (channel_mapping[i] == 255) {
+      continue;
+    }
+
+    // If it's not ignored, put it in the inverted mapping. But first check if
+    // we've told Opus to use another input channel for this coded channel:
+    const int coded_channel = channel_mapping[i];
+    if (coded_channels_to_input_channels[coded_channel] != kNotSet) {
+      // Coded channel `coded_channel` comes from both input channels
+      // `coded_channels_to_input_channels[coded_channel]` and `i`.
+      return false;
+    }
+
+    coded_channels_to_input_channels[coded_channel] = i;
+  }
+
+  // Check that we specified what input the encoder should use to produce
+  // every coded channel.
+  for (int i = 0; i < max_coded_channel; ++i) {
+    if (coded_channels_to_input_channels[i] == kNotSet) {
+      // Coded channel `i` has unspecified input channel.
+      return false;
+    }
+  }
+
+  if (num_channels > 255 || max_coded_channel >= 255) {
+    return false;
+  }
+  return true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h
new file mode 100644
index 0000000000..9b51246c15
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h
@@ -0,0 +1,66 @@
+/*
+ *  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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+struct RTC_EXPORT AudioEncoderMultiChannelOpusConfig {
+  static constexpr int kDefaultFrameSizeMs = 20;
+
+  // Opus API allows a min bitrate of 500bps, but Opus documentation suggests
+  // bitrate should be in the range of 6000 to 510000, inclusive.
+  static constexpr int kMinBitrateBps = 6000;
+  static constexpr int kMaxBitrateBps = 510000;
+
+  AudioEncoderMultiChannelOpusConfig();
+  AudioEncoderMultiChannelOpusConfig(const AudioEncoderMultiChannelOpusConfig&);
+  ~AudioEncoderMultiChannelOpusConfig();
+  AudioEncoderMultiChannelOpusConfig& operator=(
+      const AudioEncoderMultiChannelOpusConfig&);
+
+  int frame_size_ms;
+  size_t num_channels;
+  enum class ApplicationMode { kVoip, kAudio };
+  ApplicationMode application;
+  int bitrate_bps;
+  bool fec_enabled;
+  bool cbr_enabled;
+  bool dtx_enabled;
+  int max_playback_rate_hz;
+  std::vector<int> supported_frame_lengths_ms;
+
+  int complexity;
+
+  // Number of mono/stereo Opus streams.
+  int num_streams;
+
+  // Number of channel pairs coupled together, see RFC 7845 section
+  // 5.1.1. Has to be less than the number of streams
+  int coupled_streams;
+
+  // Channel mapping table, defines the mapping from encoded streams to input
+  // channels. See RFC 7845 section 5.1.1.
+  std::vector<unsigned char> channel_mapping;
+
+  bool IsOk() const;
+};
+
+}  // namespace webrtc
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multiopus_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multiopus_gn/moz.build
new file mode 100644
index 0000000000..91afd0a4e4
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multiopus_gn/moz.build
@@ -0,0 +1,226 @@
+# 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",
+    "/media/libopus/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus.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_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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_encoder_multiopus_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc
new file mode 100644
index 0000000000..5b6322da4c
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc
@@ -0,0 +1,44 @@
+/*
+ *  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 "api/audio_codecs/opus/audio_encoder_opus.h"
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+
+namespace webrtc {
+
+absl::optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
+    const SdpAudioFormat& format) {
+  return AudioEncoderOpusImpl::SdpToConfig(format);
+}
+
+void AudioEncoderOpus::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  AudioEncoderOpusImpl::AppendSupportedEncoders(specs);
+}
+
+AudioCodecInfo AudioEncoderOpus::QueryAudioEncoder(
+    const AudioEncoderOpusConfig& config) {
+  return AudioEncoderOpusImpl::QueryAudioEncoder(config);
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder(
+    const AudioEncoderOpusConfig& config,
+    int payload_type,
+    absl::optional<AudioCodecPairId> /*codec_pair_id*/,
+    const FieldTrialsView* field_trials) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return AudioEncoderOpusImpl::MakeAudioEncoder(config, payload_type);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.h
new file mode 100644
index 0000000000..df93ae5303
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.h
@@ -0,0 +1,44 @@
+/*
+ *  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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "api/field_trials_view.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// Opus encoder API for use as a template parameter to
+// CreateAudioEncoderFactory<...>().
+struct RTC_EXPORT AudioEncoderOpus {
+  using Config = AudioEncoderOpusConfig;
+  static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
+      const SdpAudioFormat& audio_format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderOpusConfig& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr);
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
new file mode 100644
index 0000000000..a9ab924b38
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.cc
@@ -0,0 +1,75 @@
+/*
+ *  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 "api/audio_codecs/opus/audio_encoder_opus_config.h"
+
+namespace webrtc {
+
+namespace {
+
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
+constexpr int kDefaultComplexity = 5;
+#else
+constexpr int kDefaultComplexity = 9;
+#endif
+
+constexpr int kDefaultLowRateComplexity =
+    WEBRTC_OPUS_VARIABLE_COMPLEXITY ? 9 : kDefaultComplexity;
+
+}  // namespace
+
+constexpr int AudioEncoderOpusConfig::kDefaultFrameSizeMs;
+constexpr int AudioEncoderOpusConfig::kMinBitrateBps;
+constexpr int AudioEncoderOpusConfig::kMaxBitrateBps;
+
+AudioEncoderOpusConfig::AudioEncoderOpusConfig()
+    : frame_size_ms(kDefaultFrameSizeMs),
+      sample_rate_hz(48000),
+      num_channels(1),
+      application(ApplicationMode::kVoip),
+      bitrate_bps(32000),
+      fec_enabled(false),
+      cbr_enabled(false),
+      max_playback_rate_hz(48000),
+      complexity(kDefaultComplexity),
+      low_rate_complexity(kDefaultLowRateComplexity),
+      complexity_threshold_bps(12500),
+      complexity_threshold_window_bps(1500),
+      dtx_enabled(false),
+      uplink_bandwidth_update_interval_ms(200),
+      payload_type(-1) {}
+AudioEncoderOpusConfig::AudioEncoderOpusConfig(const AudioEncoderOpusConfig&) =
+    default;
+AudioEncoderOpusConfig::~AudioEncoderOpusConfig() = default;
+AudioEncoderOpusConfig& AudioEncoderOpusConfig::operator=(
+    const AudioEncoderOpusConfig&) = default;
+
+bool AudioEncoderOpusConfig::IsOk() const {
+  if (frame_size_ms <= 0 || frame_size_ms % 10 != 0)
+    return false;
+  if (sample_rate_hz != 16000 && sample_rate_hz != 48000) {
+    // Unsupported input sample rate. (libopus supports a few other rates as
+    // well; we can add support for them when needed.)
+    return false;
+  }
+  if (num_channels >= 255) {
+    return false;
+  }
+  if (!bitrate_bps)
+    return false;
+  if (*bitrate_bps < kMinBitrateBps || *bitrate_bps > kMaxBitrateBps)
+    return false;
+  if (complexity < 0 || complexity > 10)
+    return false;
+  if (low_rate_complexity < 0 || low_rate_complexity > 10)
+    return false;
+  return true;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
new file mode 100644
index 0000000000..d5d7256c70
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
@@ -0,0 +1,74 @@
+/*
+ *  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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+struct RTC_EXPORT AudioEncoderOpusConfig {
+  static constexpr int kDefaultFrameSizeMs = 20;
+
+  // Opus API allows a min bitrate of 500bps, but Opus documentation suggests
+  // bitrate should be in the range of 6000 to 510000, inclusive.
+  static constexpr int kMinBitrateBps = 6000;
+  static constexpr int kMaxBitrateBps = 510000;
+
+  AudioEncoderOpusConfig();
+  AudioEncoderOpusConfig(const AudioEncoderOpusConfig&);
+  ~AudioEncoderOpusConfig();
+  AudioEncoderOpusConfig& operator=(const AudioEncoderOpusConfig&);
+
+  bool IsOk() const;  // Checks if the values are currently OK.
+
+  int frame_size_ms;
+  int sample_rate_hz;
+  size_t num_channels;
+  enum class ApplicationMode { kVoip, kAudio };
+  ApplicationMode application;
+
+  // NOTE: This member must always be set.
+  // TODO(kwiberg): Turn it into just an int.
+  absl::optional<int> bitrate_bps;
+
+  bool fec_enabled;
+  bool cbr_enabled;
+  int max_playback_rate_hz;
+
+  // `complexity` is used when the bitrate goes above
+  // `complexity_threshold_bps` + `complexity_threshold_window_bps`;
+  // `low_rate_complexity` is used when the bitrate falls below
+  // `complexity_threshold_bps` - `complexity_threshold_window_bps`. In the
+  // interval in the middle, we keep using the most recent of the two
+  // complexity settings.
+  int complexity;
+  int low_rate_complexity;
+  int complexity_threshold_bps;
+  int complexity_threshold_window_bps;
+
+  bool dtx_enabled;
+  std::vector<int> supported_frame_lengths_ms;
+  int uplink_bandwidth_update_interval_ms;
+
+  // NOTE: This member isn't necessary, and will soon go away. See
+  // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+  int payload_type;
+};
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config_gn/moz.build
new file mode 100644
index 0000000000..06732b48f4
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_config_gn/moz.build
@@ -0,0 +1,222 @@
+# 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_OPUS_VARIABLE_COMPLEXITY"] = "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/api/audio_codecs/opus/audio_encoder_opus_config.cc"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.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_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+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_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
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = 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["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
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_encoder_opus_config_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_gn/moz.build b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_gn/moz.build
new file mode 100644
index 0000000000..ab84d3f755
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus_gn/moz.build
@@ -0,0 +1,233 @@
+# 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",
+    "/media/libopus/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.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_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_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_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["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["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("audio_encoder_opus_gn")
diff --git a/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.cc b/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.cc
new file mode 100644
index 0000000000..ed68f2584e
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.cc
@@ -0,0 +1,49 @@
+/*
+ *  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 "api/audio_codecs/opus_audio_decoder_factory.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory_template.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+
+namespace webrtc {
+
+namespace {
+
+// Modify an audio decoder to not advertise support for anything.
+template <typename T>
+struct NotAdvertised {
+  using Config = typename T::Config;
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    return T::SdpToConfig(audio_format);
+  }
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
+    // Don't advertise support for anything.
+  }
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config& config,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
+    return T::MakeAudioDecoder(config, codec_pair_id);
+  }
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioDecoderFactory> CreateOpusAudioDecoderFactory() {
+  return CreateAudioDecoderFactory<
+      AudioDecoderOpus, NotAdvertised<AudioDecoderMultiChannelOpus>>();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.h b/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.h
new file mode 100644
index 0000000000..b4f497f8ff
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus_audio_decoder_factory.h
@@ -0,0 +1,26 @@
+/*
+ *  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 API_AUDIO_CODECS_OPUS_AUDIO_DECODER_FACTORY_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_DECODER_FACTORY_H_
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create only Opus audio decoders. Works like
+// CreateAudioDecoderFactory<AudioDecoderOpus>(), but is easier to use and is
+// not inline because it isn't a template.
+rtc::scoped_refptr<AudioDecoderFactory> CreateOpusAudioDecoderFactory();
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_DECODER_FACTORY_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.cc b/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.cc
new file mode 100644
index 0000000000..8c286f21e1
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.cc
@@ -0,0 +1,54 @@
+/*
+ *  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 "api/audio_codecs/opus_audio_encoder_factory.h"
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_encoder_factory_template.h"
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+
+namespace webrtc {
+namespace {
+
+// Modify an audio encoder to not advertise support for anything.
+template <typename T>
+struct NotAdvertised {
+  using Config = typename T::Config;
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    return T::SdpToConfig(audio_format);
+  }
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
+    // Don't advertise support for anything.
+  }
+  static AudioCodecInfo QueryAudioEncoder(const Config& config) {
+    return T::QueryAudioEncoder(config);
+  }
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config& config,
+      int payload_type,
+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
+      const FieldTrialsView* field_trials = nullptr) {
+    return T::MakeAudioEncoder(config, payload_type, codec_pair_id,
+                               field_trials);
+  }
+};
+
+}  // namespace
+
+rtc::scoped_refptr<AudioEncoderFactory> CreateOpusAudioEncoderFactory() {
+  return CreateAudioEncoderFactory<
+      AudioEncoderOpus, NotAdvertised<AudioEncoderMultiChannelOpus>>();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.h b/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.h
new file mode 100644
index 0000000000..8c1683b6f5
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/opus_audio_encoder_factory.h
@@ -0,0 +1,26 @@
+/*
+ *  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 API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_FACTORY_H_
+#define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_FACTORY_H_
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// Creates a new factory that can create only Opus audio encoders. Works like
+// CreateAudioEncoderFactory<AudioEncoderOpus>(), but is easier to use and is
+// not inline because it isn't a template.
+rtc::scoped_refptr<AudioEncoderFactory> CreateOpusAudioEncoderFactory();
+
+}  // namespace webrtc
+
+#endif  // API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_FACTORY_H_
diff --git a/third_party/libwebrtc/api/audio_codecs/test/BUILD.gn b/third_party/libwebrtc/api/audio_codecs/test/BUILD.gn
new file mode 100644
index 0000000000..89f5fef1ea
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/test/BUILD.gn
@@ -0,0 +1,39 @@
+# 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.
+
+import("../../../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
+if (rtc_include_tests) {
+  rtc_library("audio_codecs_api_unittests") {
+    testonly = true
+    sources = [
+      "audio_decoder_factory_template_unittest.cc",
+      "audio_encoder_factory_template_unittest.cc",
+    ]
+    deps = [
+      "..:audio_codecs_api",
+      "../../../test:audio_codec_mocks",
+      "../../../test:scoped_key_value_config",
+      "../../../test:test_support",
+      "../L16:audio_decoder_L16",
+      "../L16:audio_encoder_L16",
+      "../g711:audio_decoder_g711",
+      "../g711:audio_encoder_g711",
+      "../g722:audio_decoder_g722",
+      "../g722:audio_encoder_g722",
+      "../ilbc:audio_decoder_ilbc",
+      "../ilbc:audio_encoder_ilbc",
+      "../opus:audio_decoder_opus",
+      "../opus:audio_encoder_opus",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc b/third_party/libwebrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
new file mode 100644
index 0000000000..0b18cf934a
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
@@ -0,0 +1,222 @@
+/*
+ *  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 "api/audio_codecs/audio_decoder_factory_template.h"
+
+#include <memory>
+
+#include "api/audio_codecs/L16/audio_decoder_L16.h"
+#include "api/audio_codecs/g711/audio_decoder_g711.h"
+#include "api/audio_codecs/g722/audio_decoder_g722.h"
+#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+
+namespace {
+
+struct BogusParams {
+  static SdpAudioFormat AudioFormat() { return {"bogus", 8000, 1}; }
+  static AudioCodecInfo CodecInfo() { return {8000, 1, 12345}; }
+};
+
+struct ShamParams {
+  static SdpAudioFormat AudioFormat() {
+    return {"sham", 16000, 2, {{"param", "value"}}};
+  }
+  static AudioCodecInfo CodecInfo() { return {16000, 2, 23456}; }
+};
+
+template <typename Params>
+struct AudioDecoderFakeApi {
+  struct Config {
+    SdpAudioFormat audio_format;
+  };
+
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    if (Params::AudioFormat() == audio_format) {
+      Config config = {audio_format};
+      return config;
+    } else {
+      return absl::nullopt;
+    }
+  }
+
+  static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
+    specs->push_back({Params::AudioFormat(), Params::CodecInfo()});
+  }
+
+  static AudioCodecInfo QueryAudioDecoder(const Config&) {
+    return Params::CodecInfo();
+  }
+
+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const Config&,
+      absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) {
+    auto dec = std::make_unique<testing::StrictMock<MockAudioDecoder>>();
+    EXPECT_CALL(*dec, SampleRateHz())
+        .WillOnce(::testing::Return(Params::CodecInfo().sample_rate_hz));
+    EXPECT_CALL(*dec, Die());
+    return std::move(dec);
+  }
+};
+
+}  // namespace
+
+TEST(AudioDecoderFactoryTemplateTest, NoDecoderTypes) {
+  test::ScopedKeyValueConfig field_trials;
+  rtc::scoped_refptr<AudioDecoderFactory> factory(
+      rtc::make_ref_counted<
+          audio_decoder_factory_template_impl::AudioDecoderFactoryT<>>(
+          &field_trials));
+  EXPECT_THAT(factory->GetSupportedDecoders(), ::testing::IsEmpty());
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTemplateTest, OneDecoderType) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderFakeApi<BogusParams>>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+  auto dec = factory->MakeAudioDecoder({"bogus", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec);
+  EXPECT_EQ(8000, dec->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTemplateTest, TwoDecoderTypes) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderFakeApi<BogusParams>,
+                                           AudioDecoderFakeApi<ShamParams>>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}},
+                  AudioCodecSpec{{"sham", 16000, 2, {{"param", "value"}}},
+                                 {16000, 2, 23456}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1}));
+  EXPECT_TRUE(
+      factory->IsSupportedDecoder({"sham", 16000, 2, {{"param", "value"}}}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+  auto dec1 = factory->MakeAudioDecoder({"bogus", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec1);
+  EXPECT_EQ(8000, dec1->SampleRateHz());
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"sham", 16000, 2}, absl::nullopt));
+  auto dec2 = factory->MakeAudioDecoder(
+      {"sham", 16000, 2, {{"param", "value"}}}, absl::nullopt);
+  ASSERT_NE(nullptr, dec2);
+  EXPECT_EQ(16000, dec2->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTemplateTest, G711) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderG711>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"PCMU", 8000, 1}, {8000, 1, 64000}},
+                  AudioCodecSpec{{"PCMA", 8000, 1}, {8000, 1, 64000}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"G711", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"PCMU", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"pcma", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"pcmu", 16000, 1}, absl::nullopt));
+  auto dec1 = factory->MakeAudioDecoder({"pcmu", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec1);
+  EXPECT_EQ(8000, dec1->SampleRateHz());
+  auto dec2 = factory->MakeAudioDecoder({"PCMA", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec2);
+  EXPECT_EQ(8000, dec2->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTemplateTest, G722) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderG722>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"G722", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+  auto dec1 = factory->MakeAudioDecoder({"G722", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec1);
+  EXPECT_EQ(16000, dec1->SampleRateHz());
+  EXPECT_EQ(1u, dec1->Channels());
+  auto dec2 = factory->MakeAudioDecoder({"G722", 8000, 2}, absl::nullopt);
+  ASSERT_NE(nullptr, dec2);
+  EXPECT_EQ(16000, dec2->SampleRateHz());
+  EXPECT_EQ(2u, dec2->Channels());
+  auto dec3 = factory->MakeAudioDecoder({"G722", 8000, 3}, absl::nullopt);
+  ASSERT_EQ(nullptr, dec3);
+}
+
+TEST(AudioDecoderFactoryTemplateTest, Ilbc) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderIlbc>();
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13300}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"ilbc", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 8000, 1}, absl::nullopt));
+  auto dec = factory->MakeAudioDecoder({"ilbc", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, dec);
+  EXPECT_EQ(8000, dec->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTemplateTest, L16) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderL16>();
+  EXPECT_THAT(
+      factory->GetSupportedDecoders(),
+      ::testing::ElementsAre(
+          AudioCodecSpec{{"L16", 8000, 1}, {8000, 1, 8000 * 16}},
+          AudioCodecSpec{{"L16", 16000, 1}, {16000, 1, 16000 * 16}},
+          AudioCodecSpec{{"L16", 32000, 1}, {32000, 1, 32000 * 16}},
+          AudioCodecSpec{{"L16", 8000, 2}, {8000, 2, 8000 * 16 * 2}},
+          AudioCodecSpec{{"L16", 16000, 2}, {16000, 2, 16000 * 16 * 2}},
+          AudioCodecSpec{{"L16", 32000, 2}, {32000, 2, 32000 * 16 * 2}}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"L16", 48000, 1}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"L16", 96000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"L16", 8000, 0}, absl::nullopt));
+  auto dec = factory->MakeAudioDecoder({"L16", 48000, 2}, absl::nullopt);
+  ASSERT_NE(nullptr, dec);
+  EXPECT_EQ(48000, dec->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTemplateTest, Opus) {
+  auto factory = CreateAudioDecoderFactory<AudioDecoderOpus>();
+  AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
+  opus_info.allow_comfort_noise = false;
+  opus_info.supports_network_adaption = true;
+  const SdpAudioFormat opus_format(
+      {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}});
+  EXPECT_THAT(factory->GetSupportedDecoders(),
+              ::testing::ElementsAre(AudioCodecSpec{opus_format, opus_info}));
+  EXPECT_FALSE(factory->IsSupportedDecoder({"opus", 48000, 1}));
+  EXPECT_TRUE(factory->IsSupportedDecoder({"opus", 48000, 2}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
+  auto dec = factory->MakeAudioDecoder({"opus", 48000, 2}, absl::nullopt);
+  ASSERT_NE(nullptr, dec);
+  EXPECT_EQ(48000, dec->SampleRateHz());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc b/third_party/libwebrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
new file mode 100644
index 0000000000..dbba387724
--- /dev/null
+++ b/third_party/libwebrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
@@ -0,0 +1,224 @@
+/*
+ *  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 "api/audio_codecs/audio_encoder_factory_template.h"
+
+#include <memory>
+
+#include "api/audio_codecs/L16/audio_encoder_L16.h"
+#include "api/audio_codecs/g711/audio_encoder_g711.h"
+#include "api/audio_codecs/g722/audio_encoder_g722.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+
+namespace {
+
+struct BogusParams {
+  static SdpAudioFormat AudioFormat() { return {"bogus", 8000, 1}; }
+  static AudioCodecInfo CodecInfo() { return {8000, 1, 12345}; }
+};
+
+struct ShamParams {
+  static SdpAudioFormat AudioFormat() {
+    return {"sham", 16000, 2, {{"param", "value"}}};
+  }
+  static AudioCodecInfo CodecInfo() { return {16000, 2, 23456}; }
+};
+
+template <typename Params>
+struct AudioEncoderFakeApi {
+  struct Config {
+    SdpAudioFormat audio_format;
+  };
+
+  static absl::optional<Config> SdpToConfig(
+      const SdpAudioFormat& audio_format) {
+    if (Params::AudioFormat() == audio_format) {
+      Config config = {audio_format};
+      return config;
+    } else {
+      return absl::nullopt;
+    }
+  }
+
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
+    specs->push_back({Params::AudioFormat(), Params::CodecInfo()});
+  }
+
+  static AudioCodecInfo QueryAudioEncoder(const Config&) {
+    return Params::CodecInfo();
+  }
+
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const Config&,
+      int payload_type,
+      absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) {
+    auto enc = std::make_unique<testing::StrictMock<MockAudioEncoder>>();
+    EXPECT_CALL(*enc, SampleRateHz())
+        .WillOnce(::testing::Return(Params::CodecInfo().sample_rate_hz));
+    return std::move(enc);
+  }
+};
+
+}  // namespace
+
+TEST(AudioEncoderFactoryTemplateTest, NoEncoderTypes) {
+  test::ScopedKeyValueConfig field_trials;
+  rtc::scoped_refptr<AudioEncoderFactory> factory(
+      rtc::make_ref_counted<
+          audio_encoder_factory_template_impl::AudioEncoderFactoryT<>>(
+          &field_trials));
+  EXPECT_THAT(factory->GetSupportedEncoders(), ::testing::IsEmpty());
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+}
+
+TEST(AudioEncoderFactoryTemplateTest, OneEncoderType) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderFakeApi<BogusParams>>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(AudioCodecInfo(8000, 1, 12345),
+            factory->QueryAudioEncoder({"bogus", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+  auto enc = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc);
+  EXPECT_EQ(8000, enc->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, TwoEncoderTypes) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderFakeApi<BogusParams>,
+                                           AudioEncoderFakeApi<ShamParams>>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}},
+                  AudioCodecSpec{{"sham", 16000, 2, {{"param", "value"}}},
+                                 {16000, 2, 23456}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(AudioCodecInfo(8000, 1, 12345),
+            factory->QueryAudioEncoder({"bogus", 8000, 1}));
+  EXPECT_EQ(
+      AudioCodecInfo(16000, 2, 23456),
+      factory->QueryAudioEncoder({"sham", 16000, 2, {{"param", "value"}}}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+  auto enc1 = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc1);
+  EXPECT_EQ(8000, enc1->SampleRateHz());
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"sham", 16000, 2}, absl::nullopt));
+  auto enc2 = factory->MakeAudioEncoder(
+      17, {"sham", 16000, 2, {{"param", "value"}}}, absl::nullopt);
+  ASSERT_NE(nullptr, enc2);
+  EXPECT_EQ(16000, enc2->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, G711) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderG711>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"PCMU", 8000, 1}, {8000, 1, 64000}},
+                  AudioCodecSpec{{"PCMA", 8000, 1}, {8000, 1, 64000}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"PCMA", 16000, 1}));
+  EXPECT_EQ(AudioCodecInfo(8000, 1, 64000),
+            factory->QueryAudioEncoder({"PCMA", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"PCMU", 16000, 1}, absl::nullopt));
+  auto enc1 = factory->MakeAudioEncoder(17, {"PCMU", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc1);
+  EXPECT_EQ(8000, enc1->SampleRateHz());
+  auto enc2 = factory->MakeAudioEncoder(17, {"PCMA", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc2);
+  EXPECT_EQ(8000, enc2->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, G722) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderG722>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(AudioCodecInfo(16000, 1, 64000),
+            factory->QueryAudioEncoder({"G722", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+  auto enc = factory->MakeAudioEncoder(17, {"G722", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc);
+  EXPECT_EQ(16000, enc->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, Ilbc) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderIlbc>();
+  EXPECT_THAT(factory->GetSupportedEncoders(),
+              ::testing::ElementsAre(
+                  AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13333}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(AudioCodecInfo(8000, 1, 13333),
+            factory->QueryAudioEncoder({"ilbc", 8000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 8000, 1}, absl::nullopt));
+  auto enc = factory->MakeAudioEncoder(17, {"ilbc", 8000, 1}, absl::nullopt);
+  ASSERT_NE(nullptr, enc);
+  EXPECT_EQ(8000, enc->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, L16) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderL16>();
+  EXPECT_THAT(
+      factory->GetSupportedEncoders(),
+      ::testing::ElementsAre(
+          AudioCodecSpec{{"L16", 8000, 1}, {8000, 1, 8000 * 16}},
+          AudioCodecSpec{{"L16", 16000, 1}, {16000, 1, 16000 * 16}},
+          AudioCodecSpec{{"L16", 32000, 1}, {32000, 1, 32000 * 16}},
+          AudioCodecSpec{{"L16", 8000, 2}, {8000, 2, 8000 * 16 * 2}},
+          AudioCodecSpec{{"L16", 16000, 2}, {16000, 2, 16000 * 16 * 2}},
+          AudioCodecSpec{{"L16", 32000, 2}, {32000, 2, 32000 * 16 * 2}}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"L16", 8000, 0}));
+  EXPECT_EQ(AudioCodecInfo(48000, 1, 48000 * 16),
+            factory->QueryAudioEncoder({"L16", 48000, 1}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"L16", 8000, 0}, absl::nullopt));
+  auto enc = factory->MakeAudioEncoder(17, {"L16", 48000, 2}, absl::nullopt);
+  ASSERT_NE(nullptr, enc);
+  EXPECT_EQ(48000, enc->SampleRateHz());
+}
+
+TEST(AudioEncoderFactoryTemplateTest, Opus) {
+  auto factory = CreateAudioEncoderFactory<AudioEncoderOpus>();
+  AudioCodecInfo info = {48000, 1, 32000, 6000, 510000};
+  info.allow_comfort_noise = false;
+  info.supports_network_adaption = true;
+  EXPECT_THAT(
+      factory->GetSupportedEncoders(),
+      ::testing::ElementsAre(AudioCodecSpec{
+          {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}},
+          info}));
+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
+  EXPECT_EQ(
+      info,
+      factory->QueryAudioEncoder(
+          {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}));
+  EXPECT_EQ(nullptr,
+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
+  auto enc = factory->MakeAudioEncoder(17, {"opus", 48000, 2}, absl::nullopt);
+  ASSERT_NE(nullptr, enc);
+  EXPECT_EQ(48000, enc->SampleRateHz());
+}
+
+}  // namespace webrtc