Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

11 files changed, 2776 insertions, 0 deletions

diff --git a/third_party/libwebrtc/audio/utility/BUILD.gn b/third_party/libwebrtc/audio/utility/BUILD.gn
new file mode 100644
index 0000000000..983b6286e4
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/BUILD.gn
@@ -0,0 +1,56 @@
+# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+import("../../webrtc.gni")
+
+group("utility") {
+  deps = [ ":audio_frame_operations" ]
+}
+
+rtc_library("audio_frame_operations") {
+  visibility = [ "*" ]
+  sources = [
+    "audio_frame_operations.cc",
+    "audio_frame_operations.h",
+    "channel_mixer.cc",
+    "channel_mixer.h",
+    "channel_mixing_matrix.cc",
+    "channel_mixing_matrix.h",
+  ]
+
+  deps = [
+    "../../api/audio:audio_frame_api",
+    "../../common_audio",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:safe_conversions",
+    "../../system_wrappers:field_trial",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("utility_tests") {
+    testonly = true
+    sources = [
+      "audio_frame_operations_unittest.cc",
+      "channel_mixer_unittest.cc",
+      "channel_mixing_matrix_unittest.cc",
+    ]
+    deps = [
+      ":audio_frame_operations",
+      "../../api/audio:audio_frame_api",
+      "../../rtc_base:checks",
+      "../../rtc_base:logging",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:stringutils",
+      "../../test:field_trial",
+      "../../test:test_support",
+      "//testing/gtest",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations.cc b/third_party/libwebrtc/audio/utility/audio_frame_operations.cc
new file mode 100644
index 0000000000..1b936c239b
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/audio_frame_operations.cc
@@ -0,0 +1,294 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/audio_frame_operations.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <utility>
+
+#include "common_audio/include/audio_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace {
+
+// 2.7ms @ 48kHz, 4ms @ 32kHz, 8ms @ 16kHz.
+const size_t kMuteFadeFrames = 128;
+const float kMuteFadeInc = 1.0f / kMuteFadeFrames;
+
+}  // namespace
+
+void AudioFrameOperations::Add(const AudioFrame& frame_to_add,
+                               AudioFrame* result_frame) {
+  // Sanity check.
+  RTC_DCHECK(result_frame);
+  RTC_DCHECK_GT(result_frame->num_channels_, 0);
+  RTC_DCHECK_EQ(result_frame->num_channels_, frame_to_add.num_channels_);
+
+  bool no_previous_data = result_frame->muted();
+  if (result_frame->samples_per_channel_ != frame_to_add.samples_per_channel_) {
+    // Special case we have no data to start with.
+    RTC_DCHECK_EQ(result_frame->samples_per_channel_, 0);
+    result_frame->samples_per_channel_ = frame_to_add.samples_per_channel_;
+    no_previous_data = true;
+  }
+
+  if (result_frame->vad_activity_ == AudioFrame::kVadActive ||
+      frame_to_add.vad_activity_ == AudioFrame::kVadActive) {
+    result_frame->vad_activity_ = AudioFrame::kVadActive;
+  } else if (result_frame->vad_activity_ == AudioFrame::kVadUnknown ||
+             frame_to_add.vad_activity_ == AudioFrame::kVadUnknown) {
+    result_frame->vad_activity_ = AudioFrame::kVadUnknown;
+  }
+
+  if (result_frame->speech_type_ != frame_to_add.speech_type_)
+    result_frame->speech_type_ = AudioFrame::kUndefined;
+
+  if (!frame_to_add.muted()) {
+    const int16_t* in_data = frame_to_add.data();
+    int16_t* out_data = result_frame->mutable_data();
+    size_t length =
+        frame_to_add.samples_per_channel_ * frame_to_add.num_channels_;
+    if (no_previous_data) {
+      std::copy(in_data, in_data + length, out_data);
+    } else {
+      for (size_t i = 0; i < length; i++) {
+        const int32_t wrap_guard = static_cast<int32_t>(out_data[i]) +
+                                   static_cast<int32_t>(in_data[i]);
+        out_data[i] = rtc::saturated_cast<int16_t>(wrap_guard);
+      }
+    }
+  }
+}
+
+int AudioFrameOperations::MonoToStereo(AudioFrame* frame) {
+  if (frame->num_channels_ != 1) {
+    return -1;
+  }
+  UpmixChannels(2, frame);
+  return 0;
+}
+
+int AudioFrameOperations::StereoToMono(AudioFrame* frame) {
+  if (frame->num_channels_ != 2) {
+    return -1;
+  }
+  DownmixChannels(1, frame);
+  return frame->num_channels_ == 1 ? 0 : -1;
+}
+
+void AudioFrameOperations::QuadToStereo(const int16_t* src_audio,
+                                        size_t samples_per_channel,
+                                        int16_t* dst_audio) {
+  for (size_t i = 0; i < samples_per_channel; i++) {
+    dst_audio[i * 2] =
+        (static_cast<int32_t>(src_audio[4 * i]) + src_audio[4 * i + 1]) >> 1;
+    dst_audio[i * 2 + 1] =
+        (static_cast<int32_t>(src_audio[4 * i + 2]) + src_audio[4 * i + 3]) >>
+        1;
+  }
+}
+
+int AudioFrameOperations::QuadToStereo(AudioFrame* frame) {
+  if (frame->num_channels_ != 4) {
+    return -1;
+  }
+
+  RTC_DCHECK_LE(frame->samples_per_channel_ * 4,
+                AudioFrame::kMaxDataSizeSamples);
+
+  if (!frame->muted()) {
+    QuadToStereo(frame->data(), frame->samples_per_channel_,
+                 frame->mutable_data());
+  }
+  frame->num_channels_ = 2;
+
+  return 0;
+}
+
+void AudioFrameOperations::DownmixChannels(const int16_t* src_audio,
+                                           size_t src_channels,
+                                           size_t samples_per_channel,
+                                           size_t dst_channels,
+                                           int16_t* dst_audio) {
+  if (src_channels > 1 && dst_channels == 1) {
+    DownmixInterleavedToMono(src_audio, samples_per_channel, src_channels,
+                             dst_audio);
+    return;
+  } else if (src_channels == 4 && dst_channels == 2) {
+    QuadToStereo(src_audio, samples_per_channel, dst_audio);
+    return;
+  }
+
+  RTC_DCHECK_NOTREACHED() << "src_channels: " << src_channels
+                          << ", dst_channels: " << dst_channels;
+}
+
+void AudioFrameOperations::DownmixChannels(size_t dst_channels,
+                                           AudioFrame* frame) {
+  RTC_DCHECK_LE(frame->samples_per_channel_ * frame->num_channels_,
+                AudioFrame::kMaxDataSizeSamples);
+  if (frame->num_channels_ > 1 && dst_channels == 1) {
+    if (!frame->muted()) {
+      DownmixInterleavedToMono(frame->data(), frame->samples_per_channel_,
+                               frame->num_channels_, frame->mutable_data());
+    }
+    frame->num_channels_ = 1;
+  } else if (frame->num_channels_ == 4 && dst_channels == 2) {
+    int err = QuadToStereo(frame);
+    RTC_DCHECK_EQ(err, 0);
+  } else {
+    RTC_DCHECK_NOTREACHED() << "src_channels: " << frame->num_channels_
+                            << ", dst_channels: " << dst_channels;
+  }
+}
+
+void AudioFrameOperations::UpmixChannels(size_t target_number_of_channels,
+                                         AudioFrame* frame) {
+  RTC_DCHECK_EQ(frame->num_channels_, 1);
+  RTC_DCHECK_LE(frame->samples_per_channel_ * target_number_of_channels,
+                AudioFrame::kMaxDataSizeSamples);
+
+  if (frame->num_channels_ != 1 ||
+      frame->samples_per_channel_ * target_number_of_channels >
+          AudioFrame::kMaxDataSizeSamples) {
+    return;
+  }
+
+  if (!frame->muted()) {
+    // Up-mixing done in place. Going backwards through the frame ensure nothing
+    // is irrevocably overwritten.
+    int16_t* frame_data = frame->mutable_data();
+    for (int i = frame->samples_per_channel_ - 1; i >= 0; i--) {
+      for (size_t j = 0; j < target_number_of_channels; ++j) {
+        frame_data[target_number_of_channels * i + j] = frame_data[i];
+      }
+    }
+  }
+  frame->num_channels_ = target_number_of_channels;
+}
+
+void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  if (frame->num_channels_ != 2 || frame->muted()) {
+    return;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
+    std::swap(frame_data[i], frame_data[i + 1]);
+  }
+}
+
+void AudioFrameOperations::Mute(AudioFrame* frame,
+                                bool previous_frame_muted,
+                                bool current_frame_muted) {
+  RTC_DCHECK(frame);
+  if (!previous_frame_muted && !current_frame_muted) {
+    // Not muted, don't touch.
+  } else if (previous_frame_muted && current_frame_muted) {
+    // Frame fully muted.
+    size_t total_samples = frame->samples_per_channel_ * frame->num_channels_;
+    RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples, total_samples);
+    frame->Mute();
+  } else {
+    // Fade is a no-op on a muted frame.
+    if (frame->muted()) {
+      return;
+    }
+
+    // Limit number of samples to fade, if frame isn't long enough.
+    size_t count = kMuteFadeFrames;
+    float inc = kMuteFadeInc;
+    if (frame->samples_per_channel_ < kMuteFadeFrames) {
+      count = frame->samples_per_channel_;
+      if (count > 0) {
+        inc = 1.0f / count;
+      }
+    }
+
+    size_t start = 0;
+    size_t end = count;
+    float start_g = 0.0f;
+    if (current_frame_muted) {
+      // Fade out the last `count` samples of frame.
+      RTC_DCHECK(!previous_frame_muted);
+      start = frame->samples_per_channel_ - count;
+      end = frame->samples_per_channel_;
+      start_g = 1.0f;
+      inc = -inc;
+    } else {
+      // Fade in the first `count` samples of frame.
+      RTC_DCHECK(previous_frame_muted);
+    }
+
+    // Perform fade.
+    int16_t* frame_data = frame->mutable_data();
+    size_t channels = frame->num_channels_;
+    for (size_t j = 0; j < channels; ++j) {
+      float g = start_g;
+      for (size_t i = start * channels; i < end * channels; i += channels) {
+        g += inc;
+        frame_data[i + j] *= g;
+      }
+    }
+  }
+}
+
+void AudioFrameOperations::Mute(AudioFrame* frame) {
+  Mute(frame, true, true);
+}
+
+void AudioFrameOperations::ApplyHalfGain(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  RTC_DCHECK_GT(frame->num_channels_, 0);
+  if (frame->num_channels_ < 1 || frame->muted()) {
+    return;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       i++) {
+    frame_data[i] = frame_data[i] >> 1;
+  }
+}
+
+int AudioFrameOperations::Scale(float left, float right, AudioFrame* frame) {
+  if (frame->num_channels_ != 2) {
+    return -1;
+  } else if (frame->muted()) {
+    return 0;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_; i++) {
+    frame_data[2 * i] = static_cast<int16_t>(left * frame_data[2 * i]);
+    frame_data[2 * i + 1] = static_cast<int16_t>(right * frame_data[2 * i + 1]);
+  }
+  return 0;
+}
+
+int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame* frame) {
+  if (frame->muted()) {
+    return 0;
+  }
+
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       i++) {
+    frame_data[i] = rtc::saturated_cast<int16_t>(scale * frame_data[i]);
+  }
+  return 0;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations.h b/third_party/libwebrtc/audio/utility/audio_frame_operations.h
new file mode 100644
index 0000000000..2a5f29f4f5
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/audio_frame_operations.h
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
+#define AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "api/audio/audio_frame.h"
+
+namespace webrtc {
+
+// TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
+// Change reference parameters to pointers. Consider using a namespace rather
+// than a class.
+class AudioFrameOperations {
+ public:
+  // Add samples in `frame_to_add` with samples in `result_frame`
+  // putting the results in `results_frame`.  The fields
+  // `vad_activity_` and `speech_type_` of the result frame are
+  // updated. If `result_frame` is empty (`samples_per_channel_`==0),
+  // the samples in `frame_to_add` are added to it.  The number of
+  // channels and number of samples per channel must match except when
+  // `result_frame` is empty.
+  static void Add(const AudioFrame& frame_to_add, AudioFrame* result_frame);
+
+  // `frame.num_channels_` will be updated. This version checks for sufficient
+  // buffer size and that `num_channels_` is mono. Use UpmixChannels
+  // instead. TODO(bugs.webrtc.org/8649): remove.
+  ABSL_DEPRECATED("bugs.webrtc.org/8649")
+  static int MonoToStereo(AudioFrame* frame);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` is stereo. Use DownmixChannels
+  // instead. TODO(bugs.webrtc.org/8649): remove.
+  ABSL_DEPRECATED("bugs.webrtc.org/8649")
+  static int StereoToMono(AudioFrame* frame);
+
+  // Downmixes 4 channels `src_audio` to stereo `dst_audio`. This is an in-place
+  // operation, meaning `src_audio` and `dst_audio` may point to the same
+  // buffer.
+  static void QuadToStereo(const int16_t* src_audio,
+                           size_t samples_per_channel,
+                           int16_t* dst_audio);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` is 4 channels.
+  static int QuadToStereo(AudioFrame* frame);
+
+  // Downmixes `src_channels` `src_audio` to `dst_channels` `dst_audio`.
+  // This is an in-place operation, meaning `src_audio` and `dst_audio`
+  // may point to the same buffer. Supported channel combinations are
+  // Stereo to Mono, Quad to Mono, and Quad to Stereo.
+  static void DownmixChannels(const int16_t* src_audio,
+                              size_t src_channels,
+                              size_t samples_per_channel,
+                              size_t dst_channels,
+                              int16_t* dst_audio);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` and `dst_channels` are valid and performs relevant downmix.
+  // Supported channel combinations are N channels to Mono, and Quad to Stereo.
+  static void DownmixChannels(size_t dst_channels, AudioFrame* frame);
+
+  // `frame.num_channels_` will be updated. This version checks that
+  // `num_channels_` and `dst_channels` are valid and performs relevant
+  // downmix. Supported channel combinations are Mono to N
+  // channels. The single channel is replicated.
+  static void UpmixChannels(size_t target_number_of_channels,
+                            AudioFrame* frame);
+
+  // Swap the left and right channels of `frame`. Fails silently if `frame` is
+  // not stereo.
+  static void SwapStereoChannels(AudioFrame* frame);
+
+  // Conditionally zero out contents of `frame` for implementing audio mute:
+  //  `previous_frame_muted` &&  `current_frame_muted` - Zero out whole frame.
+  //  `previous_frame_muted` && !`current_frame_muted` - Fade-in at frame start.
+  // !`previous_frame_muted` &&  `current_frame_muted` - Fade-out at frame end.
+  // !`previous_frame_muted` && !`current_frame_muted` - Leave frame untouched.
+  static void Mute(AudioFrame* frame,
+                   bool previous_frame_muted,
+                   bool current_frame_muted);
+
+  // Zero out contents of frame.
+  static void Mute(AudioFrame* frame);
+
+  // Halve samples in `frame`.
+  static void ApplyHalfGain(AudioFrame* frame);
+
+  static int Scale(float left, float right, AudioFrame* frame);
+
+  static int ScaleWithSat(float scale, AudioFrame* frame);
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_UTILITY_AUDIO_FRAME_OPERATIONS_H_
diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations_gn/moz.build b/third_party/libwebrtc/audio/utility/audio_frame_operations_gn/moz.build
new file mode 100644
index 0000000000..81aea2d627
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/audio_frame_operations_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"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/audio/utility/audio_frame_operations.cc",
+    "/third_party/libwebrtc/audio/utility/channel_mixer.cc",
+    "/third_party/libwebrtc/audio/utility/channel_mixing_matrix.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_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_frame_operations_gn")
diff --git a/third_party/libwebrtc/audio/utility/audio_frame_operations_unittest.cc b/third_party/libwebrtc/audio/utility/audio_frame_operations_unittest.cc
new file mode 100644
index 0000000000..1a2c16e45f
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/audio_frame_operations_unittest.cc
@@ -0,0 +1,622 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/audio_frame_operations.h"
+
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+class AudioFrameOperationsTest : public ::testing::Test {
+ protected:
+  AudioFrameOperationsTest() {
+    // Set typical values.
+    frame_.samples_per_channel_ = 320;
+    frame_.num_channels_ = 2;
+  }
+
+  AudioFrame frame_;
+};
+
+class AudioFrameOperationsDeathTest : public AudioFrameOperationsTest {};
+
+void SetFrameData(int16_t ch1,
+                  int16_t ch2,
+                  int16_t ch3,
+                  int16_t ch4,
+                  AudioFrame* frame) {
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * 4; i += 4) {
+    frame_data[i] = ch1;
+    frame_data[i + 1] = ch2;
+    frame_data[i + 2] = ch3;
+    frame_data[i + 3] = ch4;
+  }
+}
+
+void SetFrameData(int16_t left, int16_t right, AudioFrame* frame) {
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) {
+    frame_data[i] = left;
+    frame_data[i + 1] = right;
+  }
+}
+
+void SetFrameData(int16_t data, AudioFrame* frame) {
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
+       i++) {
+    frame_data[i] = data;
+  }
+}
+
+void VerifyFramesAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
+  EXPECT_EQ(frame1.num_channels_, frame2.num_channels_);
+  EXPECT_EQ(frame1.samples_per_channel_, frame2.samples_per_channel_);
+  const int16_t* frame1_data = frame1.data();
+  const int16_t* frame2_data = frame2.data();
+  for (size_t i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_;
+       i++) {
+    EXPECT_EQ(frame1_data[i], frame2_data[i]);
+  }
+  EXPECT_EQ(frame1.muted(), frame2.muted());
+}
+
+void InitFrame(AudioFrame* frame,
+               size_t channels,
+               size_t samples_per_channel,
+               int16_t left_data,
+               int16_t right_data) {
+  RTC_DCHECK(frame);
+  RTC_DCHECK_GE(2, channels);
+  RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples,
+                samples_per_channel * channels);
+  frame->samples_per_channel_ = samples_per_channel;
+  frame->num_channels_ = channels;
+  if (channels == 2) {
+    SetFrameData(left_data, right_data, frame);
+  } else if (channels == 1) {
+    SetFrameData(left_data, frame);
+  }
+}
+
+int16_t GetChannelData(const AudioFrame& frame, size_t channel, size_t index) {
+  RTC_DCHECK_LT(channel, frame.num_channels_);
+  RTC_DCHECK_LT(index, frame.samples_per_channel_);
+  return frame.data()[index * frame.num_channels_ + channel];
+}
+
+void VerifyFrameDataBounds(const AudioFrame& frame,
+                           size_t channel,
+                           int16_t max,
+                           int16_t min) {
+  for (size_t i = 0; i < frame.samples_per_channel_; ++i) {
+    int16_t s = GetChannelData(frame, channel, i);
+    EXPECT_LE(min, s);
+    EXPECT_GE(max, s);
+  }
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST_F(AudioFrameOperationsDeathTest, MonoToStereoFailsWithBadParameters) {
+  EXPECT_DEATH(AudioFrameOperations::UpmixChannels(2, &frame_), "");
+  frame_.samples_per_channel_ = AudioFrame::kMaxDataSizeSamples;
+  frame_.num_channels_ = 1;
+  EXPECT_DEATH(AudioFrameOperations::UpmixChannels(2, &frame_), "");
+}
+#endif
+
+TEST_F(AudioFrameOperationsTest, MonoToStereoSucceeds) {
+  frame_.num_channels_ = 1;
+  SetFrameData(1, &frame_);
+
+  AudioFrameOperations::UpmixChannels(2, &frame_);
+  EXPECT_EQ(2u, frame_.num_channels_);
+
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = 320;
+  stereo_frame.num_channels_ = 2;
+  SetFrameData(1, 1, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, MonoToStereoMuted) {
+  frame_.num_channels_ = 1;
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::UpmixChannels(2, &frame_);
+  EXPECT_EQ(2u, frame_.num_channels_);
+  EXPECT_TRUE(frame_.muted());
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST_F(AudioFrameOperationsDeathTest, StereoToMonoFailsWithBadParameters) {
+  frame_.num_channels_ = 1;
+  EXPECT_DEATH(AudioFrameOperations::DownmixChannels(1, &frame_), "");
+}
+#endif
+
+TEST_F(AudioFrameOperationsTest, StereoToMonoSucceeds) {
+  SetFrameData(4, 2, &frame_);
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(3, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, StereoToMonoMuted) {
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, StereoToMonoBufferSucceeds) {
+  AudioFrame target_frame;
+  SetFrameData(4, 2, &frame_);
+
+  target_frame.num_channels_ = 1;
+  target_frame.samples_per_channel_ = frame_.samples_per_channel_;
+
+  AudioFrameOperations::DownmixChannels(frame_.data(), 2,
+                                        frame_.samples_per_channel_, 1,
+                                        target_frame.mutable_data());
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(3, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, target_frame);
+}
+
+TEST_F(AudioFrameOperationsTest, StereoToMonoDoesNotWrapAround) {
+  SetFrameData(-32768, -32768, &frame_);
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(-32768, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToMonoSucceeds) {
+  frame_.num_channels_ = 4;
+  SetFrameData(4, 2, 6, 8, &frame_);
+
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(5, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToMonoMuted) {
+  frame_.num_channels_ = 4;
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToMonoBufferSucceeds) {
+  AudioFrame target_frame;
+  frame_.num_channels_ = 4;
+  SetFrameData(4, 2, 6, 8, &frame_);
+
+  target_frame.num_channels_ = 1;
+  target_frame.samples_per_channel_ = frame_.samples_per_channel_;
+
+  AudioFrameOperations::DownmixChannels(frame_.data(), 4,
+                                        frame_.samples_per_channel_, 1,
+                                        target_frame.mutable_data());
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(5, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, target_frame);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToMonoDoesNotWrapAround) {
+  frame_.num_channels_ = 4;
+  SetFrameData(-32768, -32768, -32768, -32768, &frame_);
+  AudioFrameOperations::DownmixChannels(1, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels_);
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = 320;
+  mono_frame.num_channels_ = 1;
+  SetFrameData(-32768, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToStereoFailsWithBadParameters) {
+  frame_.num_channels_ = 1;
+  EXPECT_EQ(-1, AudioFrameOperations::QuadToStereo(&frame_));
+  frame_.num_channels_ = 2;
+  EXPECT_EQ(-1, AudioFrameOperations::QuadToStereo(&frame_));
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToStereoSucceeds) {
+  frame_.num_channels_ = 4;
+  SetFrameData(4, 2, 6, 8, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::QuadToStereo(&frame_));
+
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = 320;
+  stereo_frame.num_channels_ = 2;
+  SetFrameData(3, 7, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToStereoMuted) {
+  frame_.num_channels_ = 4;
+  ASSERT_TRUE(frame_.muted());
+  EXPECT_EQ(0, AudioFrameOperations::QuadToStereo(&frame_));
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToStereoBufferSucceeds) {
+  AudioFrame target_frame;
+  frame_.num_channels_ = 4;
+  SetFrameData(4, 2, 6, 8, &frame_);
+
+  target_frame.num_channels_ = 2;
+  target_frame.samples_per_channel_ = frame_.samples_per_channel_;
+
+  AudioFrameOperations::QuadToStereo(frame_.data(), frame_.samples_per_channel_,
+                                     target_frame.mutable_data());
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = 320;
+  stereo_frame.num_channels_ = 2;
+  SetFrameData(3, 7, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, target_frame);
+}
+
+TEST_F(AudioFrameOperationsTest, QuadToStereoDoesNotWrapAround) {
+  frame_.num_channels_ = 4;
+  SetFrameData(-32768, -32768, -32768, -32768, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::QuadToStereo(&frame_));
+
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = 320;
+  stereo_frame.num_channels_ = 2;
+  SetFrameData(-32768, -32768, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, SwapStereoChannelsSucceedsOnStereo) {
+  SetFrameData(0, 1, &frame_);
+
+  AudioFrame swapped_frame;
+  swapped_frame.samples_per_channel_ = 320;
+  swapped_frame.num_channels_ = 2;
+  SetFrameData(1, 0, &swapped_frame);
+
+  AudioFrameOperations::SwapStereoChannels(&frame_);
+  VerifyFramesAreEqual(swapped_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, SwapStereoChannelsMuted) {
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::SwapStereoChannels(&frame_);
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, SwapStereoChannelsFailsOnMono) {
+  frame_.num_channels_ = 1;
+  // Set data to "stereo", despite it being a mono frame.
+  SetFrameData(0, 1, &frame_);
+
+  AudioFrame orig_frame;
+  orig_frame.CopyFrom(frame_);
+  AudioFrameOperations::SwapStereoChannels(&frame_);
+  // Verify that no swap occurred.
+  VerifyFramesAreEqual(orig_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, MuteDisabled) {
+  SetFrameData(1000, -1000, &frame_);
+  AudioFrameOperations::Mute(&frame_, false, false);
+
+  AudioFrame muted_frame;
+  muted_frame.samples_per_channel_ = 320;
+  muted_frame.num_channels_ = 2;
+  SetFrameData(1000, -1000, &muted_frame);
+  VerifyFramesAreEqual(muted_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, MuteEnabled) {
+  SetFrameData(1000, -1000, &frame_);
+  AudioFrameOperations::Mute(&frame_, true, true);
+
+  AudioFrame muted_frame;
+  muted_frame.samples_per_channel_ = frame_.samples_per_channel_;
+  muted_frame.num_channels_ = frame_.num_channels_;
+  ASSERT_TRUE(muted_frame.muted());
+  VerifyFramesAreEqual(muted_frame, frame_);
+}
+
+// Verify that *beginning* to mute works for short and long (>128) frames, mono
+// and stereo. Beginning mute should yield a ramp down to zero.
+TEST_F(AudioFrameOperationsTest, MuteBeginMonoLong) {
+  InitFrame(&frame_, 1, 228, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, false, true);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 99));
+  EXPECT_EQ(992, GetChannelData(frame_, 0, 100));
+  EXPECT_EQ(7, GetChannelData(frame_, 0, 226));
+  EXPECT_EQ(0, GetChannelData(frame_, 0, 227));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteBeginMonoShort) {
+  InitFrame(&frame_, 1, 93, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, false, true);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  EXPECT_EQ(989, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(978, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(10, GetChannelData(frame_, 0, 91));
+  EXPECT_EQ(0, GetChannelData(frame_, 0, 92));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteBeginStereoLong) {
+  InitFrame(&frame_, 2, 228, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, false, true);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  VerifyFrameDataBounds(frame_, 1, 0, -1000);
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 99));
+  EXPECT_EQ(-1000, GetChannelData(frame_, 1, 99));
+  EXPECT_EQ(992, GetChannelData(frame_, 0, 100));
+  EXPECT_EQ(-992, GetChannelData(frame_, 1, 100));
+  EXPECT_EQ(7, GetChannelData(frame_, 0, 226));
+  EXPECT_EQ(-7, GetChannelData(frame_, 1, 226));
+  EXPECT_EQ(0, GetChannelData(frame_, 0, 227));
+  EXPECT_EQ(0, GetChannelData(frame_, 1, 227));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteBeginStereoShort) {
+  InitFrame(&frame_, 2, 93, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, false, true);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  VerifyFrameDataBounds(frame_, 1, 0, -1000);
+  EXPECT_EQ(989, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(-989, GetChannelData(frame_, 1, 0));
+  EXPECT_EQ(978, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(-978, GetChannelData(frame_, 1, 1));
+  EXPECT_EQ(10, GetChannelData(frame_, 0, 91));
+  EXPECT_EQ(-10, GetChannelData(frame_, 1, 91));
+  EXPECT_EQ(0, GetChannelData(frame_, 0, 92));
+  EXPECT_EQ(0, GetChannelData(frame_, 1, 92));
+}
+
+// Verify that *ending* to mute works for short and long (>128) frames, mono
+// and stereo. Ending mute should yield a ramp up from zero.
+TEST_F(AudioFrameOperationsTest, MuteEndMonoLong) {
+  InitFrame(&frame_, 1, 228, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, true, false);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  EXPECT_EQ(7, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(15, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 127));
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 128));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteEndMonoShort) {
+  InitFrame(&frame_, 1, 93, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, true, false);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  EXPECT_EQ(10, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(21, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(989, GetChannelData(frame_, 0, 91));
+  EXPECT_EQ(999, GetChannelData(frame_, 0, 92));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteEndStereoLong) {
+  InitFrame(&frame_, 2, 228, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, true, false);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  VerifyFrameDataBounds(frame_, 1, 0, -1000);
+  EXPECT_EQ(7, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(-7, GetChannelData(frame_, 1, 0));
+  EXPECT_EQ(15, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(-15, GetChannelData(frame_, 1, 1));
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 127));
+  EXPECT_EQ(-1000, GetChannelData(frame_, 1, 127));
+  EXPECT_EQ(1000, GetChannelData(frame_, 0, 128));
+  EXPECT_EQ(-1000, GetChannelData(frame_, 1, 128));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteEndStereoShort) {
+  InitFrame(&frame_, 2, 93, 1000, -1000);
+  AudioFrameOperations::Mute(&frame_, true, false);
+  VerifyFrameDataBounds(frame_, 0, 1000, 0);
+  VerifyFrameDataBounds(frame_, 1, 0, -1000);
+  EXPECT_EQ(10, GetChannelData(frame_, 0, 0));
+  EXPECT_EQ(-10, GetChannelData(frame_, 1, 0));
+  EXPECT_EQ(21, GetChannelData(frame_, 0, 1));
+  EXPECT_EQ(-21, GetChannelData(frame_, 1, 1));
+  EXPECT_EQ(989, GetChannelData(frame_, 0, 91));
+  EXPECT_EQ(-989, GetChannelData(frame_, 1, 91));
+  EXPECT_EQ(999, GetChannelData(frame_, 0, 92));
+  EXPECT_EQ(-999, GetChannelData(frame_, 1, 92));
+}
+
+TEST_F(AudioFrameOperationsTest, MuteBeginAlreadyMuted) {
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::Mute(&frame_, false, true);
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, MuteEndAlreadyMuted) {
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::Mute(&frame_, true, false);
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, ApplyHalfGainSucceeds) {
+  SetFrameData(2, &frame_);
+
+  AudioFrame half_gain_frame;
+  half_gain_frame.num_channels_ = frame_.num_channels_;
+  half_gain_frame.samples_per_channel_ = frame_.samples_per_channel_;
+  SetFrameData(1, &half_gain_frame);
+
+  AudioFrameOperations::ApplyHalfGain(&frame_);
+  VerifyFramesAreEqual(half_gain_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, ApplyHalfGainMuted) {
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::ApplyHalfGain(&frame_);
+  EXPECT_TRUE(frame_.muted());
+}
+
+// TODO(andrew): should not allow negative scales.
+TEST_F(AudioFrameOperationsTest, DISABLED_ScaleFailsWithBadParameters) {
+  frame_.num_channels_ = 1;
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, 1.0, &frame_));
+
+  frame_.num_channels_ = 3;
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, 1.0, &frame_));
+
+  frame_.num_channels_ = 2;
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(-1.0, 1.0, &frame_));
+  EXPECT_EQ(-1, AudioFrameOperations::Scale(1.0, -1.0, &frame_));
+}
+
+// TODO(andrew): fix the wraparound bug. We should always saturate.
+TEST_F(AudioFrameOperationsTest, DISABLED_ScaleDoesNotWrapAround) {
+  SetFrameData(4000, -4000, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::Scale(10.0, 10.0, &frame_));
+
+  AudioFrame clipped_frame;
+  clipped_frame.samples_per_channel_ = 320;
+  clipped_frame.num_channels_ = 2;
+  SetFrameData(32767, -32768, &clipped_frame);
+  VerifyFramesAreEqual(clipped_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, ScaleSucceeds) {
+  SetFrameData(1, -1, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::Scale(2.0, 3.0, &frame_));
+
+  AudioFrame scaled_frame;
+  scaled_frame.samples_per_channel_ = 320;
+  scaled_frame.num_channels_ = 2;
+  SetFrameData(2, -3, &scaled_frame);
+  VerifyFramesAreEqual(scaled_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, ScaleMuted) {
+  ASSERT_TRUE(frame_.muted());
+  EXPECT_EQ(0, AudioFrameOperations::Scale(2.0, 3.0, &frame_));
+  EXPECT_TRUE(frame_.muted());
+}
+
+// TODO(andrew): should fail with a negative scale.
+TEST_F(AudioFrameOperationsTest, DISABLED_ScaleWithSatFailsWithBadParameters) {
+  EXPECT_EQ(-1, AudioFrameOperations::ScaleWithSat(-1.0, &frame_));
+}
+
+TEST_F(AudioFrameOperationsTest, ScaleWithSatDoesNotWrapAround) {
+  frame_.num_channels_ = 1;
+  SetFrameData(4000, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(10.0, &frame_));
+
+  AudioFrame clipped_frame;
+  clipped_frame.samples_per_channel_ = 320;
+  clipped_frame.num_channels_ = 1;
+  SetFrameData(32767, &clipped_frame);
+  VerifyFramesAreEqual(clipped_frame, frame_);
+
+  SetFrameData(-4000, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(10.0, &frame_));
+  SetFrameData(-32768, &clipped_frame);
+  VerifyFramesAreEqual(clipped_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, ScaleWithSatSucceeds) {
+  frame_.num_channels_ = 1;
+  SetFrameData(1, &frame_);
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(2.0, &frame_));
+
+  AudioFrame scaled_frame;
+  scaled_frame.samples_per_channel_ = 320;
+  scaled_frame.num_channels_ = 1;
+  SetFrameData(2, &scaled_frame);
+  VerifyFramesAreEqual(scaled_frame, frame_);
+}
+
+TEST_F(AudioFrameOperationsTest, ScaleWithSatMuted) {
+  ASSERT_TRUE(frame_.muted());
+  EXPECT_EQ(0, AudioFrameOperations::ScaleWithSat(2.0, &frame_));
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(AudioFrameOperationsTest, AddingXToEmptyGivesX) {
+  // When samples_per_channel_ is 0, the frame counts as empty and zero.
+  AudioFrame frame_to_add_to;
+  frame_to_add_to.mutable_data();  // Unmute the frame.
+  ASSERT_FALSE(frame_to_add_to.muted());
+  frame_to_add_to.samples_per_channel_ = 0;
+  frame_to_add_to.num_channels_ = frame_.num_channels_;
+
+  SetFrameData(1000, &frame_);
+  AudioFrameOperations::Add(frame_, &frame_to_add_to);
+  VerifyFramesAreEqual(frame_, frame_to_add_to);
+}
+
+TEST_F(AudioFrameOperationsTest, AddingXToMutedGivesX) {
+  AudioFrame frame_to_add_to;
+  ASSERT_TRUE(frame_to_add_to.muted());
+  frame_to_add_to.samples_per_channel_ = frame_.samples_per_channel_;
+  frame_to_add_to.num_channels_ = frame_.num_channels_;
+
+  SetFrameData(1000, &frame_);
+  AudioFrameOperations::Add(frame_, &frame_to_add_to);
+  VerifyFramesAreEqual(frame_, frame_to_add_to);
+}
+
+TEST_F(AudioFrameOperationsTest, AddingMutedToXGivesX) {
+  AudioFrame frame_to_add_to;
+  frame_to_add_to.samples_per_channel_ = frame_.samples_per_channel_;
+  frame_to_add_to.num_channels_ = frame_.num_channels_;
+  SetFrameData(1000, &frame_to_add_to);
+
+  AudioFrame frame_copy;
+  frame_copy.CopyFrom(frame_to_add_to);
+
+  ASSERT_TRUE(frame_.muted());
+  AudioFrameOperations::Add(frame_, &frame_to_add_to);
+  VerifyFramesAreEqual(frame_copy, frame_to_add_to);
+}
+
+TEST_F(AudioFrameOperationsTest, AddingTwoFramesProducesTheirSum) {
+  AudioFrame frame_to_add_to;
+  frame_to_add_to.samples_per_channel_ = frame_.samples_per_channel_;
+  frame_to_add_to.num_channels_ = frame_.num_channels_;
+  SetFrameData(1000, &frame_to_add_to);
+  SetFrameData(2000, &frame_);
+
+  AudioFrameOperations::Add(frame_, &frame_to_add_to);
+  SetFrameData(frame_.data()[0] + 1000, &frame_);
+  VerifyFramesAreEqual(frame_, frame_to_add_to);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/channel_mixer.cc b/third_party/libwebrtc/audio/utility/channel_mixer.cc
new file mode 100644
index 0000000000..0f1e663873
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixer.cc
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/channel_mixer.h"
+
+#include "audio/utility/channel_mixing_matrix.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+ChannelMixer::ChannelMixer(ChannelLayout input_layout,
+                           ChannelLayout output_layout)
+    : input_layout_(input_layout),
+      output_layout_(output_layout),
+      input_channels_(ChannelLayoutToChannelCount(input_layout)),
+      output_channels_(ChannelLayoutToChannelCount(output_layout)) {
+  // Create the transformation matrix.
+  ChannelMixingMatrix matrix_builder(input_layout_, input_channels_,
+                                     output_layout_, output_channels_);
+  remapping_ = matrix_builder.CreateTransformationMatrix(&matrix_);
+}
+
+ChannelMixer::~ChannelMixer() = default;
+
+void ChannelMixer::Transform(AudioFrame* frame) {
+  RTC_DCHECK(frame);
+  RTC_DCHECK_EQ(matrix_[0].size(), static_cast<size_t>(input_channels_));
+  RTC_DCHECK_EQ(matrix_.size(), static_cast<size_t>(output_channels_));
+
+  // Leave the audio frame intact if the channel layouts for in and out are
+  // identical.
+  if (input_layout_ == output_layout_) {
+    return;
+  }
+
+  if (IsUpMixing()) {
+    RTC_CHECK_LE(frame->samples_per_channel() * output_channels_,
+                 frame->max_16bit_samples());
+  }
+
+  // Only change the number of output channels if the audio frame is muted.
+  if (frame->muted()) {
+    frame->num_channels_ = output_channels_;
+    frame->channel_layout_ = output_layout_;
+    return;
+  }
+
+  const int16_t* in_audio = frame->data();
+
+  // Only allocate fresh memory at first access or if the required size has
+  // increased.
+  // TODO(henrika): we might be able to do downmixing in-place and thereby avoid
+  // extra memory allocation and a memcpy.
+  const size_t num_elements = frame->samples_per_channel() * output_channels_;
+  if (audio_vector_ == nullptr || num_elements > audio_vector_size_) {
+    audio_vector_.reset(new int16_t[num_elements]);
+    audio_vector_size_ = num_elements;
+  }
+  int16_t* out_audio = audio_vector_.get();
+
+  // Modify the number of channels by creating a weighted sum of input samples
+  // where the weights (scale factors) for each output sample are given by the
+  // transformation matrix.
+  for (size_t i = 0; i < frame->samples_per_channel(); i++) {
+    for (size_t output_ch = 0; output_ch < output_channels_; ++output_ch) {
+      float acc_value = 0.0f;
+      for (size_t input_ch = 0; input_ch < input_channels_; ++input_ch) {
+        const float scale = matrix_[output_ch][input_ch];
+        // Scale should always be positive.
+        RTC_DCHECK_GE(scale, 0);
+        // Each output sample is a weighted sum of input samples.
+        acc_value += scale * in_audio[i * input_channels_ + input_ch];
+      }
+      const size_t index = output_channels_ * i + output_ch;
+      RTC_CHECK_LE(index, audio_vector_size_);
+      out_audio[index] = rtc::saturated_cast<int16_t>(acc_value);
+    }
+  }
+
+  // Update channel information.
+  frame->num_channels_ = output_channels_;
+  frame->channel_layout_ = output_layout_;
+
+  // Copy the output result to the audio frame in `frame`.
+  memcpy(
+      frame->mutable_data(), out_audio,
+      sizeof(int16_t) * frame->samples_per_channel() * frame->num_channels());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/channel_mixer.h b/third_party/libwebrtc/audio/utility/channel_mixer.h
new file mode 100644
index 0000000000..2dea8eb45b
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixer.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_UTILITY_CHANNEL_MIXER_H_
+#define AUDIO_UTILITY_CHANNEL_MIXER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio/channel_layout.h"
+
+namespace webrtc {
+
+// ChannelMixer is for converting audio between channel layouts.  The conversion
+// matrix is built upon construction and used during each Transform() call.  The
+// algorithm works by generating a conversion matrix mapping each output channel
+// to list of input channels.  The transform renders all of the output channels,
+// with each output channel rendered according to a weighted sum of the relevant
+// input channels as defined in the matrix.
+// This file is derived from Chromium's media/base/channel_mixer.h.
+class ChannelMixer {
+ public:
+  // To mix two channels into one and preserve loudness, we must apply
+  // (1 / sqrt(2)) gain to each.
+  static constexpr float kHalfPower = 0.707106781186547524401f;
+
+  ChannelMixer(ChannelLayout input_layout, ChannelLayout output_layout);
+  ~ChannelMixer();
+
+  // Transforms all input channels corresponding to the selected `input_layout`
+  // to the number of channels in the selected `output_layout`.
+  // Example usage (downmix from stereo to mono):
+  //
+  //   ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_MONO);
+  //   AudioFrame frame;
+  //   frame.samples_per_channel_ = 160;
+  //   frame.num_channels_ = 2;
+  //   EXPECT_EQ(2u, frame.channels());
+  //   mixer.Transform(&frame);
+  //   EXPECT_EQ(1u, frame.channels());
+  //
+  void Transform(AudioFrame* frame);
+
+ private:
+  bool IsUpMixing() const { return output_channels_ > input_channels_; }
+
+  // Selected channel layouts.
+  const ChannelLayout input_layout_;
+  const ChannelLayout output_layout_;
+
+  // Channel counts for input and output.
+  const size_t input_channels_;
+  const size_t output_channels_;
+
+  // 2D matrix of output channels to input channels.
+  std::vector<std::vector<float> > matrix_;
+
+  // 1D array used as temporary storage during the transformation.
+  std::unique_ptr<int16_t[]> audio_vector_;
+
+  // Number of elements allocated for `audio_vector_`.
+  size_t audio_vector_size_ = 0;
+
+  // Optimization case for when we can simply remap the input channels to output
+  // channels, i.e., when all scaling factors in `matrix_` equals 1.0.
+  bool remapping_;
+
+  // Delete the copy constructor and assignment operator.
+  ChannelMixer(const ChannelMixer& other) = delete;
+  ChannelMixer& operator=(const ChannelMixer& other) = delete;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_UTILITY_CHANNEL_MIXER_H_
diff --git a/third_party/libwebrtc/audio/utility/channel_mixer_unittest.cc b/third_party/libwebrtc/audio/utility/channel_mixer_unittest.cc
new file mode 100644
index 0000000000..94cb1ac7e3
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixer_unittest.cc
@@ -0,0 +1,393 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/channel_mixer.h"
+
+#include <memory>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio/channel_layout.h"
+#include "audio/utility/channel_mixing_matrix.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr uint32_t kTimestamp = 27;
+constexpr int kSampleRateHz = 16000;
+constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
+
+class ChannelMixerTest : public ::testing::Test {
+ protected:
+  ChannelMixerTest() {
+    // Use 10ms audio frames by default. Don't set values yet.
+    frame_.samples_per_channel_ = kSamplesPerChannel;
+    frame_.sample_rate_hz_ = kSampleRateHz;
+    EXPECT_TRUE(frame_.muted());
+  }
+
+  virtual ~ChannelMixerTest() {}
+
+  AudioFrame frame_;
+};
+
+void SetFrameData(int16_t data, AudioFrame* frame) {
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel() * frame->num_channels();
+       i++) {
+    frame_data[i] = data;
+  }
+}
+
+void SetMonoData(int16_t center, AudioFrame* frame) {
+  frame->num_channels_ = 1;
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel(); ++i) {
+    frame_data[i] = center;
+  }
+  EXPECT_FALSE(frame->muted());
+}
+
+void SetStereoData(int16_t left, int16_t right, AudioFrame* frame) {
+  ASSERT_LE(2 * frame->samples_per_channel(), frame->max_16bit_samples());
+  frame->num_channels_ = 2;
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel() * 2; i += 2) {
+    frame_data[i] = left;
+    frame_data[i + 1] = right;
+  }
+  EXPECT_FALSE(frame->muted());
+}
+
+void SetFiveOneData(int16_t front_left,
+                    int16_t front_right,
+                    int16_t center,
+                    int16_t lfe,
+                    int16_t side_left,
+                    int16_t side_right,
+                    AudioFrame* frame) {
+  ASSERT_LE(6 * frame->samples_per_channel(), frame->max_16bit_samples());
+  frame->num_channels_ = 6;
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel() * 6; i += 6) {
+    frame_data[i] = front_left;
+    frame_data[i + 1] = front_right;
+    frame_data[i + 2] = center;
+    frame_data[i + 3] = lfe;
+    frame_data[i + 4] = side_left;
+    frame_data[i + 5] = side_right;
+  }
+  EXPECT_FALSE(frame->muted());
+}
+
+void SetSevenOneData(int16_t front_left,
+                     int16_t front_right,
+                     int16_t center,
+                     int16_t lfe,
+                     int16_t side_left,
+                     int16_t side_right,
+                     int16_t back_left,
+                     int16_t back_right,
+                     AudioFrame* frame) {
+  ASSERT_LE(8 * frame->samples_per_channel(), frame->max_16bit_samples());
+  frame->num_channels_ = 8;
+  int16_t* frame_data = frame->mutable_data();
+  for (size_t i = 0; i < frame->samples_per_channel() * 8; i += 8) {
+    frame_data[i] = front_left;
+    frame_data[i + 1] = front_right;
+    frame_data[i + 2] = center;
+    frame_data[i + 3] = lfe;
+    frame_data[i + 4] = side_left;
+    frame_data[i + 5] = side_right;
+    frame_data[i + 6] = back_left;
+    frame_data[i + 7] = back_right;
+  }
+  EXPECT_FALSE(frame->muted());
+}
+
+bool AllSamplesEquals(int16_t sample, const AudioFrame* frame) {
+  const int16_t* frame_data = frame->data();
+  for (size_t i = 0; i < frame->samples_per_channel() * frame->num_channels();
+       i++) {
+    if (frame_data[i] != sample) {
+      return false;
+    }
+  }
+  return true;
+}
+
+void VerifyFramesAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
+  EXPECT_EQ(frame1.num_channels(), frame2.num_channels());
+  EXPECT_EQ(frame1.samples_per_channel(), frame2.samples_per_channel());
+  const int16_t* frame1_data = frame1.data();
+  const int16_t* frame2_data = frame2.data();
+  for (size_t i = 0; i < frame1.samples_per_channel() * frame1.num_channels();
+       i++) {
+    EXPECT_EQ(frame1_data[i], frame2_data[i]);
+  }
+  EXPECT_EQ(frame1.muted(), frame2.muted());
+}
+
+}  // namespace
+
+// Test all possible layout conversions can be constructed and mixed. Don't
+// care about the actual content, simply run through all mixing combinations
+// and ensure that nothing fails.
+TEST_F(ChannelMixerTest, ConstructAllPossibleLayouts) {
+  for (ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+       input_layout <= CHANNEL_LAYOUT_MAX;
+       input_layout = static_cast<ChannelLayout>(input_layout + 1)) {
+    for (ChannelLayout output_layout = CHANNEL_LAYOUT_MONO;
+         output_layout <= CHANNEL_LAYOUT_MAX;
+         output_layout = static_cast<ChannelLayout>(output_layout + 1)) {
+      // DISCRETE, BITSTREAM can't be tested here based on the current approach.
+      // CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC is not mixable.
+      // Stereo down mix should never be the output layout.
+      if (input_layout == CHANNEL_LAYOUT_BITSTREAM ||
+          input_layout == CHANNEL_LAYOUT_DISCRETE ||
+          input_layout == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC ||
+          output_layout == CHANNEL_LAYOUT_BITSTREAM ||
+          output_layout == CHANNEL_LAYOUT_DISCRETE ||
+          output_layout == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC ||
+          output_layout == CHANNEL_LAYOUT_STEREO_DOWNMIX) {
+        continue;
+      }
+
+      rtc::StringBuilder ss;
+      ss << "Input Layout: " << input_layout
+         << ", Output Layout: " << output_layout;
+      SCOPED_TRACE(ss.str());
+      ChannelMixer mixer(input_layout, output_layout);
+
+      frame_.UpdateFrame(kTimestamp, nullptr, kSamplesPerChannel, kSampleRateHz,
+                         AudioFrame::kNormalSpeech, AudioFrame::kVadActive,
+                         ChannelLayoutToChannelCount(input_layout));
+      EXPECT_TRUE(frame_.muted());
+      mixer.Transform(&frame_);
+    }
+  }
+}
+
+// Ensure that the audio frame is untouched when input and output channel
+// layouts are identical, i.e., the transformation should have no effect.
+// Exclude invalid mixing combinations.
+TEST_F(ChannelMixerTest, NoMixingForIdenticalChannelLayouts) {
+  for (ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+       input_layout <= CHANNEL_LAYOUT_MAX;
+       input_layout = static_cast<ChannelLayout>(input_layout + 1)) {
+    for (ChannelLayout output_layout = CHANNEL_LAYOUT_MONO;
+         output_layout <= CHANNEL_LAYOUT_MAX;
+         output_layout = static_cast<ChannelLayout>(output_layout + 1)) {
+      if (input_layout != output_layout ||
+          input_layout == CHANNEL_LAYOUT_BITSTREAM ||
+          input_layout == CHANNEL_LAYOUT_DISCRETE ||
+          input_layout == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC ||
+          output_layout == CHANNEL_LAYOUT_STEREO_DOWNMIX) {
+        continue;
+      }
+      ChannelMixer mixer(input_layout, output_layout);
+      frame_.num_channels_ = ChannelLayoutToChannelCount(input_layout);
+      SetFrameData(99, &frame_);
+      mixer.Transform(&frame_);
+      EXPECT_EQ(ChannelLayoutToChannelCount(input_layout),
+                static_cast<int>(frame_.num_channels()));
+      EXPECT_TRUE(AllSamplesEquals(99, &frame_));
+    }
+  }
+}
+
+TEST_F(ChannelMixerTest, StereoToMono) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_MONO);
+  //
+  //                      Input: stereo
+  //                      LEFT  RIGHT
+  // Output: mono CENTER  0.5   0.5
+  //
+  SetStereoData(7, 3, &frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame_.channel_layout());
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetMonoData(5, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+
+  SetStereoData(-32768, -32768, &frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame_.channel_layout());
+  SetMonoData(-32768, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(ChannelMixerTest, StereoToMonoMuted) {
+  ASSERT_TRUE(frame_.muted());
+  ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_MONO);
+  mixer.Transform(&frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame_.channel_layout());
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(ChannelMixerTest, FiveOneToSevenOneMuted) {
+  ASSERT_TRUE(frame_.muted());
+  ChannelMixer mixer(CHANNEL_LAYOUT_5_1, CHANNEL_LAYOUT_7_1);
+  mixer.Transform(&frame_);
+  EXPECT_EQ(8u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_7_1, frame_.channel_layout());
+  EXPECT_TRUE(frame_.muted());
+}
+
+TEST_F(ChannelMixerTest, FiveOneToMono) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_5_1, CHANNEL_LAYOUT_MONO);
+  //
+  //                      Input: 5.1
+  //                      LEFT   RIGHT  CENTER  LFE    SIDE_LEFT  SIDE_RIGHT
+  // Output: mono CENTER  0.707  0.707  1       0.707  0.707      0.707
+  //
+  // a = [10, 20, 15, 2, 5, 5]
+  // b = [1/sqrt(2), 1/sqrt(2), 1.0, 1/sqrt(2), 1/sqrt(2), 1/sqrt(2)] =>
+  // a * b (dot product) = 44.69848480983499,
+  // which is truncated into 44 using 16 bit representation.
+  //
+  SetFiveOneData(10, 20, 15, 2, 5, 5, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame_.channel_layout());
+
+  AudioFrame mono_frame;
+  mono_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetMonoData(44, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+
+  SetFiveOneData(-32768, -32768, -32768, -32768, -32768, -32768, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_MONO, frame_.channel_layout());
+  SetMonoData(-32768, &mono_frame);
+  VerifyFramesAreEqual(mono_frame, frame_);
+}
+
+TEST_F(ChannelMixerTest, FiveOneToSevenOne) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_5_1, CHANNEL_LAYOUT_7_1);
+  //
+  //                        Input: 5.1
+  //                        LEFT   RIGHT  CENTER  LFE    SIDE_LEFT  SIDE_RIGHT
+  // Output: 7.1 LEFT       1      0      0       0      0          0
+  //             RIGHT      0      1      0       0      0          0
+  //             CENTER     0      0      1       0      0          0
+  //             LFE        0      0      0       1      0          0
+  //             SIDE_LEFT  0      0      0       0      1          0
+  //             SIDE_RIGHT 0      0      0       0      0          1
+  //             BACK_LEFT  0      0      0       0      0          0
+  //             BACK_RIGHT 0      0      0       0      0          0
+  //
+  SetFiveOneData(10, 20, 15, 2, 5, 5, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(8u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_7_1, frame_.channel_layout());
+
+  AudioFrame seven_one_frame;
+  seven_one_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetSevenOneData(10, 20, 15, 2, 5, 5, 0, 0, &seven_one_frame);
+  VerifyFramesAreEqual(seven_one_frame, frame_);
+
+  SetFiveOneData(-32768, 32767, -32768, 32767, -32768, 32767, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(8u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_7_1, frame_.channel_layout());
+  SetSevenOneData(-32768, 32767, -32768, 32767, -32768, 32767, 0, 0,
+                  &seven_one_frame);
+  VerifyFramesAreEqual(seven_one_frame, frame_);
+}
+
+TEST_F(ChannelMixerTest, FiveOneBackToStereo) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_5_1_BACK, CHANNEL_LAYOUT_STEREO);
+  //
+  //                      Input: 5.1
+  //                      LEFT   RIGHT  CENTER  LFE    BACK_LEFT  BACK_RIGHT
+  // Output: stereo LEFT  1      0      0.707   0.707  0.707      0
+  //                RIGHT 0      1      0.707   0.707  0          0.707
+  //
+  SetFiveOneData(20, 30, 15, 2, 5, 5, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame_.channel_layout());
+
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetStereoData(35, 45, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+
+  SetFiveOneData(-32768, -32768, -32768, -32768, -32768, -32768, &frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame_.channel_layout());
+  SetStereoData(-32768, -32768, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+}
+
+TEST_F(ChannelMixerTest, MonoToStereo) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_MONO, CHANNEL_LAYOUT_STEREO);
+  //
+  //                       Input: mono
+  //                       CENTER
+  // Output: stereo LEFT   1
+  //                RIGHT  1
+  //
+  SetMonoData(44, &frame_);
+  EXPECT_EQ(1u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_STEREO, frame_.channel_layout());
+
+  AudioFrame stereo_frame;
+  stereo_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetStereoData(44, 44, &stereo_frame);
+  VerifyFramesAreEqual(stereo_frame, frame_);
+}
+
+TEST_F(ChannelMixerTest, StereoToFiveOne) {
+  ChannelMixer mixer(CHANNEL_LAYOUT_STEREO, CHANNEL_LAYOUT_5_1);
+  //
+  //                         Input: Stereo
+  //                         LEFT   RIGHT
+  // Output: 5.1 LEFT        1      0
+  //             RIGHT       0      1
+  //             CENTER      0      0
+  //             LFE         0      0
+  //             SIDE_LEFT   0      0
+  //             SIDE_RIGHT  0      0
+  //
+  SetStereoData(50, 60, &frame_);
+  EXPECT_EQ(2u, frame_.num_channels());
+  mixer.Transform(&frame_);
+  EXPECT_EQ(6u, frame_.num_channels());
+  EXPECT_EQ(CHANNEL_LAYOUT_5_1, frame_.channel_layout());
+
+  AudioFrame five_one_frame;
+  five_one_frame.samples_per_channel_ = frame_.samples_per_channel();
+  SetFiveOneData(50, 60, 0, 0, 0, 0, &five_one_frame);
+  VerifyFramesAreEqual(five_one_frame, frame_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/channel_mixing_matrix.cc b/third_party/libwebrtc/audio/utility/channel_mixing_matrix.cc
new file mode 100644
index 0000000000..1244653f63
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixing_matrix.cc
@@ -0,0 +1,333 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/channel_mixing_matrix.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+
+#include "audio/utility/channel_mixer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+// Selects the default usage of VoIP channel mapping adjustments.
+bool UseChannelMappingAdjustmentsByDefault() {
+  return !field_trial::IsEnabled(
+      "WebRTC-VoIPChannelRemixingAdjustmentKillSwitch");
+}
+
+}  // namespace
+
+static void ValidateLayout(ChannelLayout layout) {
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_NONE);
+  RTC_CHECK_LE(layout, CHANNEL_LAYOUT_MAX);
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_UNSUPPORTED);
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_DISCRETE);
+  RTC_CHECK_NE(layout, CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC);
+
+  // Verify there's at least one channel.  Should always be true here by virtue
+  // of not being one of the invalid layouts, but lets double check to be sure.
+  int channel_count = ChannelLayoutToChannelCount(layout);
+  RTC_DCHECK_GT(channel_count, 0);
+
+  // If we have more than one channel, verify a symmetric layout for sanity.
+  // The unit test will verify all possible layouts, so this can be a DCHECK.
+  // Symmetry allows simplifying the matrix building code by allowing us to
+  // assume that if one channel of a pair exists, the other will too.
+  if (channel_count > 1) {
+    // Assert that LEFT exists if and only if RIGHT exists, and so on.
+    RTC_DCHECK_EQ(ChannelOrder(layout, LEFT) >= 0,
+                  ChannelOrder(layout, RIGHT) >= 0);
+    RTC_DCHECK_EQ(ChannelOrder(layout, SIDE_LEFT) >= 0,
+                  ChannelOrder(layout, SIDE_RIGHT) >= 0);
+    RTC_DCHECK_EQ(ChannelOrder(layout, BACK_LEFT) >= 0,
+                  ChannelOrder(layout, BACK_RIGHT) >= 0);
+    RTC_DCHECK_EQ(ChannelOrder(layout, LEFT_OF_CENTER) >= 0,
+                  ChannelOrder(layout, RIGHT_OF_CENTER) >= 0);
+  } else {
+    RTC_DCHECK_EQ(layout, CHANNEL_LAYOUT_MONO);
+  }
+}
+
+ChannelMixingMatrix::ChannelMixingMatrix(ChannelLayout input_layout,
+                                         int input_channels,
+                                         ChannelLayout output_layout,
+                                         int output_channels)
+    : use_voip_channel_mapping_adjustments_(
+          UseChannelMappingAdjustmentsByDefault()),
+      input_layout_(input_layout),
+      input_channels_(input_channels),
+      output_layout_(output_layout),
+      output_channels_(output_channels) {
+  // Stereo down mix should never be the output layout.
+  RTC_CHECK_NE(output_layout, CHANNEL_LAYOUT_STEREO_DOWNMIX);
+
+  // Verify that the layouts are supported
+  if (input_layout != CHANNEL_LAYOUT_DISCRETE)
+    ValidateLayout(input_layout);
+  if (output_layout != CHANNEL_LAYOUT_DISCRETE)
+    ValidateLayout(output_layout);
+
+  // Special case for 5.0, 5.1 with back channels when upmixed to 7.0, 7.1,
+  // which should map the back LR to side LR.
+  if (input_layout_ == CHANNEL_LAYOUT_5_0_BACK &&
+      output_layout_ == CHANNEL_LAYOUT_7_0) {
+    input_layout_ = CHANNEL_LAYOUT_5_0;
+  } else if (input_layout_ == CHANNEL_LAYOUT_5_1_BACK &&
+             output_layout_ == CHANNEL_LAYOUT_7_1) {
+    input_layout_ = CHANNEL_LAYOUT_5_1;
+  }
+}
+
+ChannelMixingMatrix::~ChannelMixingMatrix() = default;
+
+bool ChannelMixingMatrix::CreateTransformationMatrix(
+    std::vector<std::vector<float>>* matrix) {
+  matrix_ = matrix;
+
+  // Size out the initial matrix.
+  matrix_->reserve(output_channels_);
+  for (int output_ch = 0; output_ch < output_channels_; ++output_ch)
+    matrix_->push_back(std::vector<float>(input_channels_, 0));
+
+  // First check for discrete case.
+  if (input_layout_ == CHANNEL_LAYOUT_DISCRETE ||
+      output_layout_ == CHANNEL_LAYOUT_DISCRETE) {
+    // If the number of input channels is more than output channels, then
+    // copy as many as we can then drop the remaining input channels.
+    // If the number of input channels is less than output channels, then
+    // copy them all, then zero out the remaining output channels.
+    int passthrough_channels = std::min(input_channels_, output_channels_);
+    for (int i = 0; i < passthrough_channels; ++i)
+      (*matrix_)[i][i] = 1;
+
+    return true;
+  }
+
+  // If specified, use adjusted channel mapping for the VoIP scenario.
+  if (use_voip_channel_mapping_adjustments_ &&
+      input_layout_ == CHANNEL_LAYOUT_MONO &&
+      ChannelLayoutToChannelCount(output_layout_) >= 2) {
+    // Only place the mono input in the front left and right channels.
+    (*matrix_)[0][0] = 1.f;
+    (*matrix_)[1][0] = 1.f;
+
+    for (size_t output_ch = 2; output_ch < matrix_->size(); ++output_ch) {
+      (*matrix_)[output_ch][0] = 0.f;
+    }
+    return true;
+  }
+
+  // Route matching channels and figure out which ones aren't accounted for.
+  for (Channels ch = LEFT; ch < CHANNELS_MAX + 1;
+       ch = static_cast<Channels>(ch + 1)) {
+    int input_ch_index = ChannelOrder(input_layout_, ch);
+    if (input_ch_index < 0)
+      continue;
+
+    int output_ch_index = ChannelOrder(output_layout_, ch);
+    if (output_ch_index < 0) {
+      unaccounted_inputs_.push_back(ch);
+      continue;
+    }
+
+    RTC_DCHECK_LT(static_cast<size_t>(output_ch_index), matrix_->size());
+    RTC_DCHECK_LT(static_cast<size_t>(input_ch_index),
+                  (*matrix_)[output_ch_index].size());
+    (*matrix_)[output_ch_index][input_ch_index] = 1;
+  }
+
+  // If all input channels are accounted for, there's nothing left to do.
+  if (unaccounted_inputs_.empty()) {
+    // Since all output channels map directly to inputs we can optimize.
+    return true;
+  }
+
+  // Mix front LR into center.
+  if (IsUnaccounted(LEFT)) {
+    // When down mixing to mono from stereo, we need to be careful of full scale
+    // stereo mixes.  Scaling by 1 / sqrt(2) here will likely lead to clipping
+    // so we use 1 / 2 instead.
+    float scale =
+        (output_layout_ == CHANNEL_LAYOUT_MONO && input_channels_ == 2)
+            ? 0.5
+            : ChannelMixer::kHalfPower;
+    Mix(LEFT, CENTER, scale);
+    Mix(RIGHT, CENTER, scale);
+  }
+
+  // Mix center into front LR.
+  if (IsUnaccounted(CENTER)) {
+    // When up mixing from mono, just do a copy to front LR.
+    float scale =
+        (input_layout_ == CHANNEL_LAYOUT_MONO) ? 1 : ChannelMixer::kHalfPower;
+    MixWithoutAccounting(CENTER, LEFT, scale);
+    Mix(CENTER, RIGHT, scale);
+  }
+
+  // Mix back LR into: side LR || back center || front LR || front center.
+  if (IsUnaccounted(BACK_LEFT)) {
+    if (HasOutputChannel(SIDE_LEFT)) {
+      // If the input has side LR, mix back LR into side LR, but instead if the
+      // input doesn't have side LR (but output does) copy back LR to side LR.
+      float scale = HasInputChannel(SIDE_LEFT) ? ChannelMixer::kHalfPower : 1;
+      Mix(BACK_LEFT, SIDE_LEFT, scale);
+      Mix(BACK_RIGHT, SIDE_RIGHT, scale);
+    } else if (HasOutputChannel(BACK_CENTER)) {
+      // Mix back LR into back center.
+      Mix(BACK_LEFT, BACK_CENTER, ChannelMixer::kHalfPower);
+      Mix(BACK_RIGHT, BACK_CENTER, ChannelMixer::kHalfPower);
+    } else if (output_layout_ > CHANNEL_LAYOUT_MONO) {
+      // Mix back LR into front LR.
+      Mix(BACK_LEFT, LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_RIGHT, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix back LR into front center.
+      Mix(BACK_LEFT, CENTER, ChannelMixer::kHalfPower);
+      Mix(BACK_RIGHT, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix side LR into: back LR || back center || front LR || front center.
+  if (IsUnaccounted(SIDE_LEFT)) {
+    if (HasOutputChannel(BACK_LEFT)) {
+      // If the input has back LR, mix side LR into back LR, but instead if the
+      // input doesn't have back LR (but output does) copy side LR to back LR.
+      float scale = HasInputChannel(BACK_LEFT) ? ChannelMixer::kHalfPower : 1;
+      Mix(SIDE_LEFT, BACK_LEFT, scale);
+      Mix(SIDE_RIGHT, BACK_RIGHT, scale);
+    } else if (HasOutputChannel(BACK_CENTER)) {
+      // Mix side LR into back center.
+      Mix(SIDE_LEFT, BACK_CENTER, ChannelMixer::kHalfPower);
+      Mix(SIDE_RIGHT, BACK_CENTER, ChannelMixer::kHalfPower);
+    } else if (output_layout_ > CHANNEL_LAYOUT_MONO) {
+      // Mix side LR into front LR.
+      Mix(SIDE_LEFT, LEFT, ChannelMixer::kHalfPower);
+      Mix(SIDE_RIGHT, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix side LR into front center.
+      Mix(SIDE_LEFT, CENTER, ChannelMixer::kHalfPower);
+      Mix(SIDE_RIGHT, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix back center into: back LR || side LR || front LR || front center.
+  if (IsUnaccounted(BACK_CENTER)) {
+    if (HasOutputChannel(BACK_LEFT)) {
+      // Mix back center into back LR.
+      MixWithoutAccounting(BACK_CENTER, BACK_LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_CENTER, BACK_RIGHT, ChannelMixer::kHalfPower);
+    } else if (HasOutputChannel(SIDE_LEFT)) {
+      // Mix back center into side LR.
+      MixWithoutAccounting(BACK_CENTER, SIDE_LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_CENTER, SIDE_RIGHT, ChannelMixer::kHalfPower);
+    } else if (output_layout_ > CHANNEL_LAYOUT_MONO) {
+      // Mix back center into front LR.
+      // TODO(dalecurtis): Not sure about these values?
+      MixWithoutAccounting(BACK_CENTER, LEFT, ChannelMixer::kHalfPower);
+      Mix(BACK_CENTER, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix back center into front center.
+      // TODO(dalecurtis): Not sure about these values?
+      Mix(BACK_CENTER, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix LR of center into: front LR || front center.
+  if (IsUnaccounted(LEFT_OF_CENTER)) {
+    if (HasOutputChannel(LEFT)) {
+      // Mix LR of center into front LR.
+      Mix(LEFT_OF_CENTER, LEFT, ChannelMixer::kHalfPower);
+      Mix(RIGHT_OF_CENTER, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix LR of center into front center.
+      Mix(LEFT_OF_CENTER, CENTER, ChannelMixer::kHalfPower);
+      Mix(RIGHT_OF_CENTER, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // Mix LFE into: front center || front LR.
+  if (IsUnaccounted(LFE)) {
+    if (!HasOutputChannel(CENTER)) {
+      // Mix LFE into front LR.
+      MixWithoutAccounting(LFE, LEFT, ChannelMixer::kHalfPower);
+      Mix(LFE, RIGHT, ChannelMixer::kHalfPower);
+    } else {
+      // Mix LFE into front center.
+      Mix(LFE, CENTER, ChannelMixer::kHalfPower);
+    }
+  }
+
+  // All channels should now be accounted for.
+  RTC_DCHECK(unaccounted_inputs_.empty());
+
+  // See if the output `matrix_` is simply a remapping matrix.  If each input
+  // channel maps to a single output channel we can simply remap.  Doing this
+  // programmatically is less fragile than logic checks on channel mappings.
+  for (int output_ch = 0; output_ch < output_channels_; ++output_ch) {
+    int input_mappings = 0;
+    for (int input_ch = 0; input_ch < input_channels_; ++input_ch) {
+      // We can only remap if each row contains a single scale of 1.  I.e., each
+      // output channel is mapped from a single unscaled input channel.
+      if ((*matrix_)[output_ch][input_ch] != 1 || ++input_mappings > 1)
+        return false;
+    }
+  }
+
+  // If we've gotten here, `matrix_` is simply a remapping.
+  return true;
+}
+
+void ChannelMixingMatrix::AccountFor(Channels ch) {
+  unaccounted_inputs_.erase(
+      std::find(unaccounted_inputs_.begin(), unaccounted_inputs_.end(), ch));
+}
+
+bool ChannelMixingMatrix::IsUnaccounted(Channels ch) const {
+  return std::find(unaccounted_inputs_.begin(), unaccounted_inputs_.end(),
+                   ch) != unaccounted_inputs_.end();
+}
+
+bool ChannelMixingMatrix::HasInputChannel(Channels ch) const {
+  return ChannelOrder(input_layout_, ch) >= 0;
+}
+
+bool ChannelMixingMatrix::HasOutputChannel(Channels ch) const {
+  return ChannelOrder(output_layout_, ch) >= 0;
+}
+
+void ChannelMixingMatrix::Mix(Channels input_ch,
+                              Channels output_ch,
+                              float scale) {
+  MixWithoutAccounting(input_ch, output_ch, scale);
+  AccountFor(input_ch);
+}
+
+void ChannelMixingMatrix::MixWithoutAccounting(Channels input_ch,
+                                               Channels output_ch,
+                                               float scale) {
+  int input_ch_index = ChannelOrder(input_layout_, input_ch);
+  int output_ch_index = ChannelOrder(output_layout_, output_ch);
+
+  RTC_DCHECK(IsUnaccounted(input_ch));
+  RTC_DCHECK_GE(input_ch_index, 0);
+  RTC_DCHECK_GE(output_ch_index, 0);
+
+  RTC_DCHECK_EQ((*matrix_)[output_ch_index][input_ch_index], 0);
+  (*matrix_)[output_ch_index][input_ch_index] = scale;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/audio/utility/channel_mixing_matrix.h b/third_party/libwebrtc/audio/utility/channel_mixing_matrix.h
new file mode 100644
index 0000000000..ee00860846
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixing_matrix.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef AUDIO_UTILITY_CHANNEL_MIXING_MATRIX_H_
+#define AUDIO_UTILITY_CHANNEL_MIXING_MATRIX_H_
+
+#include <vector>
+
+#include "api/audio/channel_layout.h"
+
+namespace webrtc {
+
+class ChannelMixingMatrix {
+ public:
+  ChannelMixingMatrix(ChannelLayout input_layout,
+                      int input_channels,
+                      ChannelLayout output_layout,
+                      int output_channels);
+
+  ~ChannelMixingMatrix();
+
+  // Create the transformation matrix of input channels to output channels.
+  // Updates the empty matrix with the transformation, and returns true
+  // if the transformation is just a remapping of channels (no mixing).
+  // The size of `matrix` is `output_channels` x `input_channels`, i.e., the
+  // number of rows equals the number of output channels and the number of
+  // columns corresponds to the number of input channels.
+  // This file is derived from Chromium's media/base/channel_mixing_matrix.h.
+  bool CreateTransformationMatrix(std::vector<std::vector<float>>* matrix);
+
+ private:
+  const bool use_voip_channel_mapping_adjustments_;
+
+  // Result transformation of input channels to output channels
+  std::vector<std::vector<float>>* matrix_;
+
+  // Input and output channel layout provided during construction.
+  ChannelLayout input_layout_;
+  int input_channels_;
+  ChannelLayout output_layout_;
+  int output_channels_;
+
+  // Helper variable for tracking which inputs are currently unaccounted,
+  // should be empty after construction completes.
+  std::vector<Channels> unaccounted_inputs_;
+
+  // Helper methods for managing unaccounted input channels.
+  void AccountFor(Channels ch);
+  bool IsUnaccounted(Channels ch) const;
+
+  // Helper methods for checking if `ch` exists in either `input_layout_` or
+  // `output_layout_` respectively.
+  bool HasInputChannel(Channels ch) const;
+  bool HasOutputChannel(Channels ch) const;
+
+  // Helper methods for updating `matrix_` with the proper value for
+  // mixing `input_ch` into `output_ch`.  MixWithoutAccounting() does not
+  // remove the channel from `unaccounted_inputs_`.
+  void Mix(Channels input_ch, Channels output_ch, float scale);
+  void MixWithoutAccounting(Channels input_ch, Channels output_ch, float scale);
+
+  // Delete the copy constructor and assignment operator.
+  ChannelMixingMatrix(const ChannelMixingMatrix& other) = delete;
+  ChannelMixingMatrix& operator=(const ChannelMixingMatrix& other) = delete;
+};
+
+}  // namespace webrtc
+
+#endif  // AUDIO_UTILITY_CHANNEL_MIXING_MATRIX_H_
diff --git a/third_party/libwebrtc/audio/utility/channel_mixing_matrix_unittest.cc b/third_party/libwebrtc/audio/utility/channel_mixing_matrix_unittest.cc
new file mode 100644
index 0000000000..a4efb4fd38
--- /dev/null
+++ b/third_party/libwebrtc/audio/utility/channel_mixing_matrix_unittest.cc
@@ -0,0 +1,476 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio/utility/channel_mixing_matrix.h"
+
+#include <stddef.h>
+
+#include "audio/utility/channel_mixer.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Test all possible layout conversions can be constructed and mixed.
+// Also ensure that the channel matrix fulfill certain conditions when remapping
+// is supported.
+TEST(ChannelMixingMatrixTest, ConstructAllPossibleLayouts) {
+  for (ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+       input_layout <= CHANNEL_LAYOUT_MAX;
+       input_layout = static_cast<ChannelLayout>(input_layout + 1)) {
+    for (ChannelLayout output_layout = CHANNEL_LAYOUT_MONO;
+         output_layout <= CHANNEL_LAYOUT_MAX;
+         output_layout = static_cast<ChannelLayout>(output_layout + 1)) {
+      // DISCRETE, BITSTREAM can't be tested here based on the current approach.
+      // CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC is not mixable.
+      // Stereo down mix should never be the output layout.
+      if (input_layout == CHANNEL_LAYOUT_BITSTREAM ||
+          input_layout == CHANNEL_LAYOUT_DISCRETE ||
+          input_layout == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC ||
+          output_layout == CHANNEL_LAYOUT_BITSTREAM ||
+          output_layout == CHANNEL_LAYOUT_DISCRETE ||
+          output_layout == CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC ||
+          output_layout == CHANNEL_LAYOUT_STEREO_DOWNMIX) {
+        continue;
+      }
+
+      rtc::StringBuilder ss;
+      ss << "Input Layout: " << input_layout
+         << ", Output Layout: " << output_layout;
+      SCOPED_TRACE(ss.str());
+      ChannelMixingMatrix matrix_builder(
+          input_layout, ChannelLayoutToChannelCount(input_layout),
+          output_layout, ChannelLayoutToChannelCount(output_layout));
+      const int input_channels = ChannelLayoutToChannelCount(input_layout);
+      const int output_channels = ChannelLayoutToChannelCount(output_layout);
+      std::vector<std::vector<float>> matrix;
+      bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+      if (remapping) {
+        // Also ensure that (when remapping can take place), a maximum of one
+        // input channel is included per output. This knowledge will simplify
+        // the channel mixing algorithm since it allows us to find the only
+        // scale factor which equals 1.0 and copy that input to its
+        // corresponding output. If no such factor can be found, the
+        // corresponding output can be set to zero.
+        for (int i = 0; i < output_channels; i++) {
+          EXPECT_EQ(static_cast<size_t>(input_channels), matrix[i].size());
+          int num_input_channels_accounted_for_per_output = 0;
+          for (int j = 0; j < input_channels; j++) {
+            float scale = matrix[i][j];
+            if (scale > 0) {
+              EXPECT_EQ(scale, 1.0f);
+              num_input_channels_accounted_for_per_output++;
+            }
+          }
+          // Each output channel shall contain contribution from one or less
+          // input channels.
+          EXPECT_LE(num_input_channels_accounted_for_per_output, 1);
+        }
+      }
+    }
+  }
+}
+
+// Verify channels are mixed and scaled correctly.
+TEST(ChannelMixingMatrixTest, StereoToMono) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_STEREO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_MONO;
+  ChannelMixingMatrix matrix_builder(
+      input_layout, ChannelLayoutToChannelCount(input_layout), output_layout,
+      ChannelLayoutToChannelCount(output_layout));
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                      Input: stereo
+  //                      LEFT  RIGHT
+  // Output: mono CENTER  0.5   0.5
+  //
+  EXPECT_FALSE(remapping);
+  EXPECT_EQ(1u, matrix.size());
+  EXPECT_EQ(2u, matrix[0].size());
+  EXPECT_EQ(0.5f, matrix[0][0]);
+  EXPECT_EQ(0.5f, matrix[0][1]);
+}
+
+TEST(ChannelMixingMatrixTest, MonoToStereo) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_STEREO;
+  ChannelMixingMatrix matrix_builder(
+      input_layout, ChannelLayoutToChannelCount(input_layout), output_layout,
+      ChannelLayoutToChannelCount(output_layout));
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                       Input: mono
+  //                       CENTER
+  // Output: stereo LEFT   1
+  //                RIGHT  1
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(2u, matrix.size());
+  EXPECT_EQ(1u, matrix[0].size());
+  EXPECT_EQ(1.0f, matrix[0][0]);
+  EXPECT_EQ(1u, matrix[1].size());
+  EXPECT_EQ(1.0f, matrix[1][0]);
+}
+
+TEST(ChannelMixingMatrixTest, MonoToTwoOneWithoutVoIPAdjustments) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VoIPChannelRemixingAdjustmentKillSwitch/Enabled/");
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_2_1;
+  ChannelMixingMatrix matrix_builder(
+      input_layout, ChannelLayoutToChannelCount(input_layout), output_layout,
+      ChannelLayoutToChannelCount(output_layout));
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                         Input: mono
+  //                         CENTER
+  // Output: 2.1 FRONT_LEFT  1
+  //             FRONT_RIGHT 1
+  //             BACK_CENTER 0
+  //
+  EXPECT_FALSE(remapping);
+  EXPECT_EQ(3u, matrix.size());
+  EXPECT_EQ(1u, matrix[0].size());
+  EXPECT_EQ(1.0f, matrix[0][0]);
+  EXPECT_EQ(1.0f, matrix[1][0]);
+  EXPECT_EQ(0.0f, matrix[2][0]);
+}
+
+TEST(ChannelMixingMatrixTest, MonoToTwoOneWithVoIPAdjustments) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_2_1;
+  ChannelMixingMatrix matrix_builder(
+      input_layout, ChannelLayoutToChannelCount(input_layout), output_layout,
+      ChannelLayoutToChannelCount(output_layout));
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                         Input: mono
+  //                         CENTER
+  // Output: 2.1 FRONT_LEFT  1
+  //             FRONT_RIGHT 1
+  //             BACK_CENTER 0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(3u, matrix.size());
+  EXPECT_EQ(1u, matrix[0].size());
+  EXPECT_EQ(1.0f, matrix[0][0]);
+  EXPECT_EQ(1.0f, matrix[1][0]);
+  EXPECT_EQ(0.0f, matrix[2][0]);
+}
+
+TEST(ChannelMixingMatrixTest, MonoToFiveOneWithoutVoIPAdjustments) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VoIPChannelRemixingAdjustmentKillSwitch/Enabled/");
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_5_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+  //                         Input: mono
+  //                         CENTER
+  // Output: 5.1 LEFT        0
+  //             RIGHT       0
+  //             CENTER      1
+  //             LFE         0
+  //             SIDE_LEFT   0
+  //             SIDE_RIGHT  0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int n = 0; n < output_channels; n++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[n].size());
+    if (n == CENTER) {
+      EXPECT_EQ(1.0f, matrix[CENTER][0]);
+    } else {
+      EXPECT_EQ(0.0f, matrix[n][0]);
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, MonoToFiveOneWithVoIPAdjustments) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_5_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+  //                         Input: mono
+  //                         CENTER
+  // Output: 5.1 LEFT        1
+  //             RIGHT       1
+  //             CENTER      0
+  //             LFE         0
+  //             SIDE_LEFT   0
+  //             SIDE_RIGHT  0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int n = 0; n < output_channels; n++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[n].size());
+    if (n == LEFT || n == RIGHT) {
+      EXPECT_EQ(1.0f, matrix[n][0]);
+    } else {
+      EXPECT_EQ(0.0f, matrix[n][0]);
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, MonoToSevenOneWithoutVoIPAdjustments) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-VoIPChannelRemixingAdjustmentKillSwitch/Enabled/");
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_7_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+  //                         Input: mono
+  //                         CENTER
+  // Output: 7.1 LEFT        0
+  //             RIGHT       0
+  //             CENTER      1
+  //             LFE         0
+  //             SIDE_LEFT   0
+  //             SIDE_RIGHT  0
+  //             BACK_LEFT   0
+  //             BACK_RIGHT  0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int n = 0; n < output_channels; n++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[n].size());
+    if (n == CENTER) {
+      EXPECT_EQ(1.0f, matrix[CENTER][0]);
+    } else {
+      EXPECT_EQ(0.0f, matrix[n][0]);
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, MonoToSevenOneWithVoIPAdjustments) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_MONO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_7_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+  //                         Input: mono
+  //                         CENTER
+  // Output: 7.1 LEFT        1
+  //             RIGHT       1
+  //             CENTER      0
+  //             LFE         0
+  //             SIDE_LEFT   0
+  //             SIDE_RIGHT  0
+  //             BACK_LEFT   0
+  //             BACK_RIGHT  0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int n = 0; n < output_channels; n++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[n].size());
+    if (n == LEFT || n == RIGHT) {
+      EXPECT_EQ(1.0f, matrix[n][0]);
+    } else {
+      EXPECT_EQ(0.0f, matrix[n][0]);
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, FiveOneToMono) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_5_1;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_MONO;
+  ChannelMixingMatrix matrix_builder(
+      input_layout, ChannelLayoutToChannelCount(input_layout), output_layout,
+      ChannelLayoutToChannelCount(output_layout));
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  // Note: 1/sqrt(2) is shown as 0.707.
+  //
+  //                      Input: 5.1
+  //                      LEFT   RIGHT  CENTER  LFE    SIDE_LEFT  SIDE_RIGHT
+  // Output: mono CENTER  0.707  0.707  1       0.707  0.707      0.707
+  //
+  EXPECT_FALSE(remapping);
+  EXPECT_EQ(1u, matrix.size());
+  EXPECT_EQ(6u, matrix[0].size());
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[0][0]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[0][1]);
+  // The center channel will be mixed at scale 1.
+  EXPECT_EQ(1.0f, matrix[0][2]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[0][3]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[0][4]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[0][5]);
+}
+
+TEST(ChannelMixingMatrixTest, FiveOneBackToStereo) {
+  // Front L, Front R, Front C, LFE, Back L, Back R
+  ChannelLayout input_layout = CHANNEL_LAYOUT_5_1_BACK;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_STEREO;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  // Note: 1/sqrt(2) is shown as 0.707.
+  // Note: The Channels enumerator is given by {LEFT = 0, RIGHT, CENTER, LFE,
+  // BACK_LEFT, BACK_RIGHT,...}, hence we can use the enumerator values as
+  // indexes in the matrix when verifying the scaling factors.
+  //
+  //                      Input: 5.1
+  //                      LEFT   RIGHT  CENTER  LFE    BACK_LEFT  BACK_RIGHT
+  // Output: stereo LEFT  1      0      0.707   0.707  0.707      0
+  //                RIGHT 0      1      0.707   0.707  0          0.707
+  //
+  EXPECT_FALSE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  EXPECT_EQ(static_cast<size_t>(input_channels), matrix[LEFT].size());
+  EXPECT_EQ(static_cast<size_t>(input_channels), matrix[RIGHT].size());
+  EXPECT_EQ(1.0f, matrix[LEFT][LEFT]);
+  EXPECT_EQ(1.0f, matrix[RIGHT][RIGHT]);
+  EXPECT_EQ(0.0f, matrix[LEFT][RIGHT]);
+  EXPECT_EQ(0.0f, matrix[RIGHT][LEFT]);
+  EXPECT_EQ(0.0f, matrix[LEFT][BACK_RIGHT]);
+  EXPECT_EQ(0.0f, matrix[RIGHT][BACK_LEFT]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[LEFT][CENTER]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[LEFT][LFE]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[LEFT][BACK_LEFT]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[RIGHT][CENTER]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[RIGHT][LFE]);
+  EXPECT_FLOAT_EQ(ChannelMixer::kHalfPower, matrix[RIGHT][BACK_RIGHT]);
+}
+
+TEST(ChannelMixingMatrixTest, FiveOneToSevenOne) {
+  // Front L, Front R, Front C, LFE, Side L, Side R
+  ChannelLayout input_layout = CHANNEL_LAYOUT_5_1;
+  // Front L, Front R, Front C, LFE, Side L, Side R, Back L, Back R
+  ChannelLayout output_layout = CHANNEL_LAYOUT_7_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                        Input: 5.1
+  //                        LEFT   RIGHT  CENTER  LFE    SIDE_LEFT  SIDE_RIGHT
+  // Output: 7.1 LEFT       1      0      0       0      0          0
+  //             RIGHT      0      1      0       0      0          0
+  //             CENTER     0      0      1       0      0          0
+  //             LFE        0      0      0       1      0          0
+  //             SIDE_LEFT  0      0      0       0      1          0
+  //             SIDE_RIGHT 0      0      0       0      0          1
+  //             BACK_LEFT  0      0      0       0      0          0
+  //             BACK_RIGHT 0      0      0       0      0          0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int i = 0; i < output_channels; i++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[i].size());
+    for (int j = 0; j < input_channels; j++) {
+      if (i == j) {
+        EXPECT_EQ(1.0f, matrix[i][j]);
+      } else {
+        EXPECT_EQ(0.0f, matrix[i][j]);
+      }
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, StereoToFiveOne) {
+  ChannelLayout input_layout = CHANNEL_LAYOUT_STEREO;
+  ChannelLayout output_layout = CHANNEL_LAYOUT_5_1;
+  const int input_channels = ChannelLayoutToChannelCount(input_layout);
+  const int output_channels = ChannelLayoutToChannelCount(output_layout);
+  ChannelMixingMatrix matrix_builder(input_layout, input_channels,
+                                     output_layout, output_channels);
+  std::vector<std::vector<float>> matrix;
+  bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+
+  //                         Input: Stereo
+  //                         LEFT   RIGHT
+  // Output: 5.1 LEFT        1      0
+  //             RIGHT       0      1
+  //             CENTER      0      0
+  //             LFE         0      0
+  //             SIDE_LEFT   0      0
+  //             SIDE_RIGHT  0      0
+  //
+  EXPECT_TRUE(remapping);
+  EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+  for (int n = 0; n < output_channels; n++) {
+    EXPECT_EQ(static_cast<size_t>(input_channels), matrix[n].size());
+    if (n == LEFT) {
+      EXPECT_EQ(1.0f, matrix[LEFT][LEFT]);
+      EXPECT_EQ(0.0f, matrix[LEFT][RIGHT]);
+    } else if (n == RIGHT) {
+      EXPECT_EQ(0.0f, matrix[RIGHT][LEFT]);
+      EXPECT_EQ(1.0f, matrix[RIGHT][RIGHT]);
+    } else {
+      EXPECT_EQ(0.0f, matrix[n][LEFT]);
+      EXPECT_EQ(0.0f, matrix[n][RIGHT]);
+    }
+  }
+}
+
+TEST(ChannelMixingMatrixTest, DiscreteToDiscrete) {
+  const struct {
+    int input_channels;
+    int output_channels;
+  } test_case[] = {
+      {2, 2},
+      {2, 5},
+      {5, 2},
+  };
+
+  for (size_t n = 0; n < arraysize(test_case); n++) {
+    int input_channels = test_case[n].input_channels;
+    int output_channels = test_case[n].output_channels;
+    ChannelMixingMatrix matrix_builder(CHANNEL_LAYOUT_DISCRETE, input_channels,
+                                       CHANNEL_LAYOUT_DISCRETE,
+                                       output_channels);
+    std::vector<std::vector<float>> matrix;
+    bool remapping = matrix_builder.CreateTransformationMatrix(&matrix);
+    EXPECT_TRUE(remapping);
+    EXPECT_EQ(static_cast<size_t>(output_channels), matrix.size());
+    for (int i = 0; i < output_channels; i++) {
+      EXPECT_EQ(static_cast<size_t>(input_channels), matrix[i].size());
+      for (int j = 0; j < input_channels; j++) {
+        if (i == j) {
+          EXPECT_EQ(1.0f, matrix[i][j]);
+        } else {
+          EXPECT_EQ(0.0f, matrix[i][j]);
+        }
+      }
+    }
+  }
+}
+
+}  // namespace webrtc