Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

233 files changed, 28644 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/audio_decoder.h b/third_party/libwebrtc/modules/audio_coding/codecs/audio_decoder.h
new file mode 100644
index 0000000000..b7b15cdd6e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/audio_decoder.h
@@ -0,0 +1,20 @@
+/*
+ *  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.
+ */
+
+// This file is for backwards compatibility only! Use
+// webrtc/api/audio_codecs/audio_decoder.h instead!
+// TODO(kwiberg): Remove it.
+
+#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
+#define MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
+
+#include "api/audio_codecs/audio_decoder.h"
+
+#endif  // MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/audio_encoder.h b/third_party/libwebrtc/modules/audio_coding/codecs/audio_encoder.h
new file mode 100644
index 0000000000..010ae6705f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -0,0 +1,20 @@
+/*
+ *  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.
+ */
+
+// This file is for backwards compatibility only! Use
+// webrtc/api/audio_codecs/audio_encoder.h instead!
+// TODO(ossu): Remove it.
+
+#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
+#define MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
+
+#include "api/audio_codecs/audio_encoder.h"
+
+#endif  // MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
new file mode 100644
index 0000000000..4a2b261a59
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+
+#include <memory>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(AudioDecoderFactoryTest, CreateUnknownDecoder) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("rey", 8000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreatePcmu) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // PCMu supports 8 kHz, and any number of channels.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 0), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 1), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 2), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 3), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 16000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreatePcma) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // PCMa supports 8 kHz, and any number of channels.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 0), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 1), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 2), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 3), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 16000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreateIlbc) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // iLBC supports 8 kHz, 1 channel.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 0), absl::nullopt));
+#ifdef WEBRTC_CODEC_ILBC
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1), absl::nullopt));
+#endif
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 2), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1), absl::nullopt));
+}
+
+TEST(AudioDecoderFactoryTest, CreateL16) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // L16 supports any clock rate and any number of channels up to 24.
+  const int clockrates[] = {8000, 16000, 32000, 48000};
+  const int num_channels[] = {1, 2, 3, 24};
+  for (int clockrate : clockrates) {
+    EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, 0),
+                                       absl::nullopt));
+    for (int channels : num_channels) {
+      EXPECT_TRUE(adf->MakeAudioDecoder(
+          SdpAudioFormat("l16", clockrate, channels), absl::nullopt));
+    }
+  }
+}
+
+// Tests that using more channels than the maximum does not work
+TEST(AudioDecoderFactoryTest, MaxNrOfChannels) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  std::vector<std::string> codecs = {
+#ifdef WEBRTC_CODEC_OPUS
+    "opus",
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+    "ilbc",
+#endif
+    "pcmu",
+    "pcma",
+    "l16",
+    "G722",
+    "G711",
+  };
+
+  for (auto codec : codecs) {
+    EXPECT_FALSE(adf->MakeAudioDecoder(
+        SdpAudioFormat(codec, 32000, AudioDecoder::kMaxNumberOfChannels + 1),
+        absl::nullopt));
+  }
+}
+
+TEST(AudioDecoderFactoryTest, CreateG722) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // g722 supports 8 kHz, 1-2 channels.
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 0), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt));
+  EXPECT_TRUE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 2), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 3), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 16000, 1), absl::nullopt));
+  EXPECT_FALSE(
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 32000, 1), absl::nullopt));
+
+  // g722 actually uses a 16 kHz sample rate instead of the nominal 8 kHz.
+  std::unique_ptr<AudioDecoder> dec =
+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt);
+  EXPECT_EQ(16000, dec->SampleRateHz());
+}
+
+TEST(AudioDecoderFactoryTest, CreateOpus) {
+  rtc::scoped_refptr<AudioDecoderFactory> adf =
+      CreateBuiltinAudioDecoderFactory();
+  ASSERT_TRUE(adf);
+  // Opus supports 48 kHz, 2 channels, and wants a "stereo" parameter whose
+  // value is either "0" or "1".
+  for (int hz : {8000, 16000, 32000, 48000}) {
+    for (int channels : {0, 1, 2, 3}) {
+      for (std::string stereo : {"XX", "0", "1", "2"}) {
+        SdpAudioFormat::Parameters params;
+        if (stereo != "XX") {
+          params["stereo"] = stereo;
+        }
+        const bool good = (hz == 48000 && channels == 2 &&
+                           (stereo == "XX" || stereo == "0" || stereo == "1"));
+        EXPECT_EQ(good,
+                  static_cast<bool>(adf->MakeAudioDecoder(
+                      SdpAudioFormat("opus", hz, channels, std::move(params)),
+                      absl::nullopt)));
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
new file mode 100644
index 0000000000..26ae1eda8a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
@@ -0,0 +1,178 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+
+#include <limits>
+#include <memory>
+#include <vector>
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class AudioEncoderFactoryTest
+    : public ::testing::TestWithParam<rtc::scoped_refptr<AudioEncoderFactory>> {
+};
+
+TEST_P(AudioEncoderFactoryTest, SupportsAtLeastOneFormat) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  EXPECT_FALSE(supported_encoders.empty());
+}
+
+TEST_P(AudioEncoderFactoryTest, CanQueryAllSupportedFormats) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto info = factory->QueryAudioEncoder(spec.format);
+    EXPECT_TRUE(info);
+  }
+}
+
+TEST_P(AudioEncoderFactoryTest, CanConstructAllSupportedEncoders) {
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto info = factory->QueryAudioEncoder(spec.format);
+    auto encoder = factory->MakeAudioEncoder(127, spec.format, absl::nullopt);
+    EXPECT_TRUE(encoder);
+    EXPECT_EQ(encoder->SampleRateHz(), info->sample_rate_hz);
+    EXPECT_EQ(encoder->NumChannels(), info->num_channels);
+    EXPECT_EQ(encoder->RtpTimestampRateHz(), spec.format.clockrate_hz);
+  }
+}
+
+TEST_P(AudioEncoderFactoryTest, CanRunAllSupportedEncoders) {
+  constexpr int kTestPayloadType = 127;
+  auto factory = GetParam();
+  auto supported_encoders = factory->GetSupportedEncoders();
+  for (const auto& spec : supported_encoders) {
+    auto encoder =
+        factory->MakeAudioEncoder(kTestPayloadType, spec.format, absl::nullopt);
+    EXPECT_TRUE(encoder);
+    encoder->Reset();
+    const int num_samples = rtc::checked_cast<int>(
+        encoder->SampleRateHz() * encoder->NumChannels() / 100);
+    rtc::Buffer out;
+    rtc::BufferT<int16_t> audio;
+    audio.SetData(num_samples, [](rtc::ArrayView<int16_t> audio) {
+      for (size_t i = 0; i != audio.size(); ++i) {
+        // Just put some numbers in there, ensure they're within range.
+        audio[i] =
+            static_cast<int16_t>(i & std::numeric_limits<int16_t>::max());
+      }
+      return audio.size();
+    });
+    // This is here to stop the test going forever with a broken encoder.
+    constexpr int kMaxEncodeCalls = 100;
+    int blocks = 0;
+    for (; blocks < kMaxEncodeCalls; ++blocks) {
+      AudioEncoder::EncodedInfo info = encoder->Encode(
+          blocks * encoder->RtpTimestampRateHz() / 100, audio, &out);
+      EXPECT_EQ(info.encoded_bytes, out.size());
+      if (info.encoded_bytes > 0) {
+        EXPECT_EQ(0u, info.encoded_timestamp);
+        EXPECT_EQ(kTestPayloadType, info.payload_type);
+        break;
+      }
+    }
+    ASSERT_LT(blocks, kMaxEncodeCalls);
+    const unsigned int next_timestamp =
+        blocks * encoder->RtpTimestampRateHz() / 100;
+    out.Clear();
+    for (; blocks < kMaxEncodeCalls; ++blocks) {
+      AudioEncoder::EncodedInfo info = encoder->Encode(
+          blocks * encoder->RtpTimestampRateHz() / 100, audio, &out);
+      EXPECT_EQ(info.encoded_bytes, out.size());
+      if (info.encoded_bytes > 0) {
+        EXPECT_EQ(next_timestamp, info.encoded_timestamp);
+        EXPECT_EQ(kTestPayloadType, info.payload_type);
+        break;
+      }
+    }
+    ASSERT_LT(blocks, kMaxEncodeCalls);
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(BuiltinAudioEncoderFactoryTest,
+                         AudioEncoderFactoryTest,
+                         ::testing::Values(CreateBuiltinAudioEncoderFactory()));
+
+TEST(BuiltinAudioEncoderFactoryTest, SupportsTheExpectedFormats) {
+  using ::testing::ElementsAreArray;
+  // Check that we claim to support the formats we expect from build flags, and
+  // we've ordered them correctly.
+  auto factory = CreateBuiltinAudioEncoderFactory();
+  auto specs = factory->GetSupportedEncoders();
+
+  const std::vector<SdpAudioFormat> supported_formats = [&specs] {
+    std::vector<SdpAudioFormat> formats;
+    formats.reserve(specs.size());
+    for (const auto& spec : specs) {
+      formats.push_back(spec.format);
+    }
+    return formats;
+  }();
+
+  const std::vector<SdpAudioFormat> expected_formats = {
+#ifdef WEBRTC_CODEC_OPUS
+    {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}},
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+    {"isac", 16000, 1},
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+    {"isac", 32000, 1},
+#endif
+    {"G722", 8000, 1},
+#ifdef WEBRTC_CODEC_ILBC
+    {"ilbc", 8000, 1},
+#endif
+    {"pcmu", 8000, 1},
+    {"pcma", 8000, 1}
+  };
+
+  ASSERT_THAT(supported_formats, ElementsAreArray(expected_formats));
+}
+
+// Tests that using more channels than the maximum does not work.
+TEST(BuiltinAudioEncoderFactoryTest, MaxNrOfChannels) {
+  rtc::scoped_refptr<AudioEncoderFactory> aef =
+      CreateBuiltinAudioEncoderFactory();
+  std::vector<std::string> codecs = {
+#ifdef WEBRTC_CODEC_OPUS
+    "opus",
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+    "isac",
+#endif
+#ifdef WEBRTC_CODEC_ILBC
+    "ilbc",
+#endif
+    "pcmu",
+    "pcma",
+    "l16",
+    "G722",
+    "G711",
+  };
+
+  for (auto codec : codecs) {
+    EXPECT_FALSE(aef->MakeAudioEncoder(
+        /*payload_type=*/111,
+        /*format=*/
+        SdpAudioFormat(codec, 32000, AudioEncoder::kMaxNumberOfChannels + 1),
+        /*codec_pair_id=*/absl::nullopt));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
new file mode 100644
index 0000000000..7546ac178f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -0,0 +1,322 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kMaxFrameSizeMs = 60;
+
+class AudioEncoderCng final : public AudioEncoder {
+ public:
+  explicit AudioEncoderCng(AudioEncoderCngConfig&& config);
+  ~AudioEncoderCng() override;
+
+  // Not copyable or moveable.
+  AudioEncoderCng(const AudioEncoderCng&) = delete;
+  AudioEncoderCng(AudioEncoderCng&&) = delete;
+  AudioEncoderCng& operator=(const AudioEncoderCng&) = delete;
+  AudioEncoderCng& operator=(AudioEncoderCng&&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+  void Reset() override;
+  bool SetFec(bool enable) override;
+  bool SetDtx(bool enable) override;
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
+      override;
+  void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) override;
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms) override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ private:
+  EncodedInfo EncodePassive(size_t frames_to_encode, rtc::Buffer* encoded);
+  EncodedInfo EncodeActive(size_t frames_to_encode, rtc::Buffer* encoded);
+  size_t SamplesPer10msFrame() const;
+
+  std::unique_ptr<AudioEncoder> speech_encoder_;
+  const int cng_payload_type_;
+  const int num_cng_coefficients_;
+  const int sid_frame_interval_ms_;
+  std::vector<int16_t> speech_buffer_;
+  std::vector<uint32_t> rtp_timestamps_;
+  bool last_frame_active_;
+  std::unique_ptr<Vad> vad_;
+  std::unique_ptr<ComfortNoiseEncoder> cng_encoder_;
+};
+
+AudioEncoderCng::AudioEncoderCng(AudioEncoderCngConfig&& config)
+    : speech_encoder_((static_cast<void>([&] {
+                         RTC_CHECK(config.IsOk()) << "Invalid configuration.";
+                       }()),
+                       std::move(config.speech_encoder))),
+      cng_payload_type_(config.payload_type),
+      num_cng_coefficients_(config.num_cng_coefficients),
+      sid_frame_interval_ms_(config.sid_frame_interval_ms),
+      last_frame_active_(true),
+      vad_(config.vad ? std::unique_ptr<Vad>(config.vad)
+                      : CreateVad(config.vad_mode)),
+      cng_encoder_(new ComfortNoiseEncoder(SampleRateHz(),
+                                           sid_frame_interval_ms_,
+                                           num_cng_coefficients_)) {}
+
+AudioEncoderCng::~AudioEncoderCng() = default;
+
+int AudioEncoderCng::SampleRateHz() const {
+  return speech_encoder_->SampleRateHz();
+}
+
+size_t AudioEncoderCng::NumChannels() const {
+  return 1;
+}
+
+int AudioEncoderCng::RtpTimestampRateHz() const {
+  return speech_encoder_->RtpTimestampRateHz();
+}
+
+size_t AudioEncoderCng::Num10MsFramesInNextPacket() const {
+  return speech_encoder_->Num10MsFramesInNextPacket();
+}
+
+size_t AudioEncoderCng::Max10MsFramesInAPacket() const {
+  return speech_encoder_->Max10MsFramesInAPacket();
+}
+
+int AudioEncoderCng::GetTargetBitrate() const {
+  return speech_encoder_->GetTargetBitrate();
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  const size_t samples_per_10ms_frame = SamplesPer10msFrame();
+  RTC_CHECK_EQ(speech_buffer_.size(),
+               rtp_timestamps_.size() * samples_per_10ms_frame);
+  rtp_timestamps_.push_back(rtp_timestamp);
+  RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
+  speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
+  const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
+  if (rtp_timestamps_.size() < frames_to_encode) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_LE(frames_to_encode * 10, kMaxFrameSizeMs)
+      << "Frame size cannot be larger than " << kMaxFrameSizeMs
+      << " ms when using VAD/CNG.";
+
+  // Group several 10 ms blocks per VAD call. Call VAD once or twice using the
+  // following split sizes:
+  // 10 ms = 10 + 0 ms; 20 ms = 20 + 0 ms; 30 ms = 30 + 0 ms;
+  // 40 ms = 20 + 20 ms; 50 ms = 30 + 20 ms; 60 ms = 30 + 30 ms.
+  size_t blocks_in_first_vad_call =
+      (frames_to_encode > 3 ? 3 : frames_to_encode);
+  if (frames_to_encode == 4)
+    blocks_in_first_vad_call = 2;
+  RTC_CHECK_GE(frames_to_encode, blocks_in_first_vad_call);
+  const size_t blocks_in_second_vad_call =
+      frames_to_encode - blocks_in_first_vad_call;
+
+  // Check if all of the buffer is passive speech. Start with checking the first
+  // block.
+  Vad::Activity activity = vad_->VoiceActivity(
+      &speech_buffer_[0], samples_per_10ms_frame * blocks_in_first_vad_call,
+      SampleRateHz());
+  if (activity == Vad::kPassive && blocks_in_second_vad_call > 0) {
+    // Only check the second block if the first was passive.
+    activity = vad_->VoiceActivity(
+        &speech_buffer_[samples_per_10ms_frame * blocks_in_first_vad_call],
+        samples_per_10ms_frame * blocks_in_second_vad_call, SampleRateHz());
+  }
+
+  EncodedInfo info;
+  switch (activity) {
+    case Vad::kPassive: {
+      info = EncodePassive(frames_to_encode, encoded);
+      last_frame_active_ = false;
+      break;
+    }
+    case Vad::kActive: {
+      info = EncodeActive(frames_to_encode, encoded);
+      last_frame_active_ = true;
+      break;
+    }
+    default: {
+      RTC_CHECK_NOTREACHED();
+    }
+  }
+
+  speech_buffer_.erase(
+      speech_buffer_.begin(),
+      speech_buffer_.begin() + frames_to_encode * samples_per_10ms_frame);
+  rtp_timestamps_.erase(rtp_timestamps_.begin(),
+                        rtp_timestamps_.begin() + frames_to_encode);
+  return info;
+}
+
+void AudioEncoderCng::Reset() {
+  speech_encoder_->Reset();
+  speech_buffer_.clear();
+  rtp_timestamps_.clear();
+  last_frame_active_ = true;
+  vad_->Reset();
+  cng_encoder_.reset(new ComfortNoiseEncoder(
+      SampleRateHz(), sid_frame_interval_ms_, num_cng_coefficients_));
+}
+
+bool AudioEncoderCng::SetFec(bool enable) {
+  return speech_encoder_->SetFec(enable);
+}
+
+bool AudioEncoderCng::SetDtx(bool enable) {
+  return speech_encoder_->SetDtx(enable);
+}
+
+bool AudioEncoderCng::SetApplication(Application application) {
+  return speech_encoder_->SetApplication(application);
+}
+
+void AudioEncoderCng::SetMaxPlaybackRate(int frequency_hz) {
+  speech_encoder_->SetMaxPlaybackRate(frequency_hz);
+}
+
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoderCng::ReclaimContainedEncoders() {
+  return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
+}
+
+void AudioEncoderCng::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  speech_encoder_->OnReceivedUplinkPacketLossFraction(
+      uplink_packet_loss_fraction);
+}
+
+void AudioEncoderCng::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {
+  speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
+                                             bwe_period_ms);
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderCng::GetFrameLengthRange() const {
+  return speech_encoder_->GetFrameLengthRange();
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
+    size_t frames_to_encode,
+    rtc::Buffer* encoded) {
+  bool force_sid = last_frame_active_;
+  bool output_produced = false;
+  const size_t samples_per_10ms_frame = SamplesPer10msFrame();
+  AudioEncoder::EncodedInfo info;
+
+  for (size_t i = 0; i < frames_to_encode; ++i) {
+    // It's important not to pass &info.encoded_bytes directly to
+    // WebRtcCng_Encode(), since later loop iterations may return zero in
+    // that value, in which case we don't want to overwrite any value from
+    // an earlier iteration.
+    size_t encoded_bytes_tmp =
+        cng_encoder_->Encode(rtc::ArrayView<const int16_t>(
+                                 &speech_buffer_[i * samples_per_10ms_frame],
+                                 samples_per_10ms_frame),
+                             force_sid, encoded);
+
+    if (encoded_bytes_tmp > 0) {
+      RTC_CHECK(!output_produced);
+      info.encoded_bytes = encoded_bytes_tmp;
+      output_produced = true;
+      force_sid = false;
+    }
+  }
+
+  info.encoded_timestamp = rtp_timestamps_.front();
+  info.payload_type = cng_payload_type_;
+  info.send_even_if_empty = true;
+  info.speech = false;
+  return info;
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(size_t frames_to_encode,
+                                                        rtc::Buffer* encoded) {
+  const size_t samples_per_10ms_frame = SamplesPer10msFrame();
+  AudioEncoder::EncodedInfo info;
+  for (size_t i = 0; i < frames_to_encode; ++i) {
+    info =
+        speech_encoder_->Encode(rtp_timestamps_.front(),
+                                rtc::ArrayView<const int16_t>(
+                                    &speech_buffer_[i * samples_per_10ms_frame],
+                                    samples_per_10ms_frame),
+                                encoded);
+    if (i + 1 == frames_to_encode) {
+      RTC_CHECK_GT(info.encoded_bytes, 0) << "Encoder didn't deliver data.";
+    } else {
+      RTC_CHECK_EQ(info.encoded_bytes, 0)
+          << "Encoder delivered data too early.";
+    }
+  }
+  return info;
+}
+
+size_t AudioEncoderCng::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
+}
+
+}  // namespace
+
+AudioEncoderCngConfig::AudioEncoderCngConfig() = default;
+AudioEncoderCngConfig::AudioEncoderCngConfig(AudioEncoderCngConfig&&) = default;
+AudioEncoderCngConfig::~AudioEncoderCngConfig() = default;
+
+bool AudioEncoderCngConfig::IsOk() const {
+  if (num_channels != 1)
+    return false;
+  if (!speech_encoder)
+    return false;
+  if (num_channels != speech_encoder->NumChannels())
+    return false;
+  if (sid_frame_interval_ms <
+      static_cast<int>(speech_encoder->Max10MsFramesInAPacket() * 10))
+    return false;
+  if (num_cng_coefficients > WEBRTC_CNG_MAX_LPC_ORDER ||
+      num_cng_coefficients <= 0)
+    return false;
+  return true;
+}
+
+std::unique_ptr<AudioEncoder> CreateComfortNoiseEncoder(
+    AudioEncoderCngConfig&& config) {
+  return std::make_unique<AudioEncoderCng>(std::move(config));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
new file mode 100644
index 0000000000..8a1183489f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+#define MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+
+#include <stddef.h>
+
+#include <memory>
+
+#include "api/audio_codecs/audio_encoder.h"
+#include "common_audio/vad/include/vad.h"
+
+namespace webrtc {
+
+struct AudioEncoderCngConfig {
+  // Moveable, not copyable.
+  AudioEncoderCngConfig();
+  AudioEncoderCngConfig(AudioEncoderCngConfig&&);
+  ~AudioEncoderCngConfig();
+
+  bool IsOk() const;
+
+  size_t num_channels = 1;
+  int payload_type = 13;
+  std::unique_ptr<AudioEncoder> speech_encoder;
+  Vad::Aggressiveness vad_mode = Vad::kVadNormal;
+  int sid_frame_interval_ms = 100;
+  int num_cng_coefficients = 8;
+  // The Vad pointer is mainly for testing. If a NULL pointer is passed, the
+  // AudioEncoderCng creates (and destroys) a Vad object internally. If an
+  // object is passed, the AudioEncoderCng assumes ownership of the Vad
+  // object.
+  Vad* vad = nullptr;
+};
+
+std::unique_ptr<AudioEncoder> CreateComfortNoiseEncoder(
+    AudioEncoderCngConfig&& config);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
new file mode 100644
index 0000000000..c688004363
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -0,0 +1,520 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+
+#include <memory>
+#include <vector>
+
+#include "common_audio/vad/mock/mock_vad.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
+#include "test/testsupport/rtc_expect_death.h"
+
+using ::testing::_;
+using ::testing::Eq;
+using ::testing::InSequence;
+using ::testing::Invoke;
+using ::testing::Not;
+using ::testing::Optional;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+namespace webrtc {
+
+namespace {
+static const size_t kMaxNumSamples = 48 * 10 * 2;  // 10 ms @ 48 kHz stereo.
+static const size_t kMockReturnEncodedBytes = 17;
+static const int kCngPayloadType = 18;
+}  // namespace
+
+class AudioEncoderCngTest : public ::testing::Test {
+ protected:
+  AudioEncoderCngTest()
+      : mock_encoder_owner_(new MockAudioEncoder),
+        mock_encoder_(mock_encoder_owner_.get()),
+        mock_vad_(new MockVad),
+        timestamp_(4711),
+        num_audio_samples_10ms_(0),
+        sample_rate_hz_(8000) {
+    memset(audio_, 0, kMaxNumSamples * 2);
+    EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(1));
+  }
+
+  AudioEncoderCngTest(const AudioEncoderCngTest&) = delete;
+  AudioEncoderCngTest& operator=(const AudioEncoderCngTest&) = delete;
+
+  void TearDown() override {
+    EXPECT_CALL(*mock_vad_, Die()).Times(1);
+    cng_.reset();
+  }
+
+  AudioEncoderCngConfig MakeCngConfig() {
+    AudioEncoderCngConfig config;
+    config.speech_encoder = std::move(mock_encoder_owner_);
+    EXPECT_TRUE(config.speech_encoder);
+
+    // Let the AudioEncoderCng object use a MockVad instead of its internally
+    // created Vad object.
+    config.vad = mock_vad_;
+    config.payload_type = kCngPayloadType;
+
+    return config;
+  }
+
+  void CreateCng(AudioEncoderCngConfig&& config) {
+    num_audio_samples_10ms_ = static_cast<size_t>(10 * sample_rate_hz_ / 1000);
+    ASSERT_LE(num_audio_samples_10ms_, kMaxNumSamples);
+    if (config.speech_encoder) {
+      EXPECT_CALL(*mock_encoder_, SampleRateHz())
+          .WillRepeatedly(Return(sample_rate_hz_));
+      // Max10MsFramesInAPacket() is just used to verify that the SID frame
+      // period is not too small. The return value does not matter that much,
+      // as long as it is smaller than 10.
+      EXPECT_CALL(*mock_encoder_, Max10MsFramesInAPacket())
+          .WillOnce(Return(1u));
+    }
+    cng_ = CreateComfortNoiseEncoder(std::move(config));
+  }
+
+  void Encode() {
+    ASSERT_TRUE(cng_) << "Must call CreateCng() first.";
+    encoded_info_ = cng_->Encode(
+        timestamp_,
+        rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms_),
+        &encoded_);
+    timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_);
+  }
+
+  // Expect `num_calls` calls to the encoder, all successful. The last call
+  // claims to have encoded `kMockReturnEncodedBytes` bytes, and all the
+  // preceding ones 0 bytes.
+  void ExpectEncodeCalls(size_t num_calls) {
+    InSequence s;
+    AudioEncoder::EncodedInfo info;
+    for (size_t j = 0; j < num_calls - 1; ++j) {
+      EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).WillOnce(Return(info));
+    }
+    info.encoded_bytes = kMockReturnEncodedBytes;
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(
+            Invoke(MockAudioEncoder::FakeEncoding(kMockReturnEncodedBytes)));
+  }
+
+  // Verifies that the cng_ object waits until it has collected
+  // `blocks_per_frame` blocks of audio, and then dispatches all of them to
+  // the underlying codec (speech or cng).
+  void CheckBlockGrouping(size_t blocks_per_frame, bool active_speech) {
+    EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+        .WillRepeatedly(Return(blocks_per_frame));
+    auto config = MakeCngConfig();
+    const int num_cng_coefficients = config.num_cng_coefficients;
+    CreateCng(std::move(config));
+    EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+        .WillRepeatedly(Return(active_speech ? Vad::kActive : Vad::kPassive));
+
+    // Don't expect any calls to the encoder yet.
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).Times(0);
+    for (size_t i = 0; i < blocks_per_frame - 1; ++i) {
+      Encode();
+      EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+    }
+    if (active_speech)
+      ExpectEncodeCalls(blocks_per_frame);
+    Encode();
+    if (active_speech) {
+      EXPECT_EQ(kMockReturnEncodedBytes, encoded_info_.encoded_bytes);
+    } else {
+      EXPECT_EQ(static_cast<size_t>(num_cng_coefficients + 1),
+                encoded_info_.encoded_bytes);
+    }
+  }
+
+  // Verifies that the audio is partitioned into larger blocks before calling
+  // the VAD.
+  void CheckVadInputSize(int input_frame_size_ms,
+                         int expected_first_block_size_ms,
+                         int expected_second_block_size_ms) {
+    const size_t blocks_per_frame =
+        static_cast<size_t>(input_frame_size_ms / 10);
+
+    EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+        .WillRepeatedly(Return(blocks_per_frame));
+
+    // Expect nothing to happen before the last block is sent to cng_.
+    EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _)).Times(0);
+    for (size_t i = 0; i < blocks_per_frame - 1; ++i) {
+      Encode();
+    }
+
+    // Let the VAD decision be passive, since an active decision may lead to
+    // early termination of the decision loop.
+    InSequence s;
+    EXPECT_CALL(
+        *mock_vad_,
+        VoiceActivity(_, expected_first_block_size_ms * sample_rate_hz_ / 1000,
+                      sample_rate_hz_))
+        .WillOnce(Return(Vad::kPassive));
+    if (expected_second_block_size_ms > 0) {
+      EXPECT_CALL(*mock_vad_,
+                  VoiceActivity(
+                      _, expected_second_block_size_ms * sample_rate_hz_ / 1000,
+                      sample_rate_hz_))
+          .WillOnce(Return(Vad::kPassive));
+    }
+
+    // With this call to Encode(), `mock_vad_` should be called according to the
+    // above expectations.
+    Encode();
+  }
+
+  // Tests a frame with both active and passive speech. Returns true if the
+  // decision was active speech, false if it was passive.
+  bool CheckMixedActivePassive(Vad::Activity first_type,
+                               Vad::Activity second_type) {
+    // Set the speech encoder frame size to 60 ms, to ensure that the VAD will
+    // be called twice.
+    const size_t blocks_per_frame = 6;
+    EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+        .WillRepeatedly(Return(blocks_per_frame));
+    InSequence s;
+    EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+        .WillOnce(Return(first_type));
+    if (first_type == Vad::kPassive) {
+      // Expect a second call to the VAD only if the first frame was passive.
+      EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+          .WillOnce(Return(second_type));
+    }
+    encoded_info_.payload_type = 0;
+    for (size_t i = 0; i < blocks_per_frame; ++i) {
+      Encode();
+    }
+    return encoded_info_.payload_type != kCngPayloadType;
+  }
+
+  std::unique_ptr<AudioEncoder> cng_;
+  std::unique_ptr<MockAudioEncoder> mock_encoder_owner_;
+  MockAudioEncoder* mock_encoder_;
+  MockVad* mock_vad_;  // Ownership is transferred to `cng_`.
+  uint32_t timestamp_;
+  int16_t audio_[kMaxNumSamples];
+  size_t num_audio_samples_10ms_;
+  rtc::Buffer encoded_;
+  AudioEncoder::EncodedInfo encoded_info_;
+  int sample_rate_hz_;
+};
+
+TEST_F(AudioEncoderCngTest, CreateAndDestroy) {
+  CreateCng(MakeCngConfig());
+}
+
+TEST_F(AudioEncoderCngTest, CheckFrameSizePropagation) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillOnce(Return(17U));
+  EXPECT_EQ(17U, cng_->Num10MsFramesInNextPacket());
+}
+
+TEST_F(AudioEncoderCngTest, CheckTargetAudioBitratePropagation) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_,
+              OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
+  cng_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
+}
+
+TEST_F(AudioEncoderCngTest, CheckPacketLossFractionPropagation) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, OnReceivedUplinkPacketLossFraction(0.5));
+  cng_->OnReceivedUplinkPacketLossFraction(0.5);
+}
+
+TEST_F(AudioEncoderCngTest, CheckGetFrameLengthRangePropagation) {
+  CreateCng(MakeCngConfig());
+  auto expected_range =
+      std::make_pair(TimeDelta::Millis(20), TimeDelta::Millis(20));
+  EXPECT_CALL(*mock_encoder_, GetFrameLengthRange())
+      .WillRepeatedly(Return(absl::make_optional(expected_range)));
+  EXPECT_THAT(cng_->GetFrameLengthRange(), Optional(Eq(expected_range)));
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCallsVad) {
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(1U));
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kPassive));
+  Encode();
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects1BlockPassiveSpeech) {
+  CheckBlockGrouping(1, false);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects2BlocksPassiveSpeech) {
+  CheckBlockGrouping(2, false);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects3BlocksPassiveSpeech) {
+  CheckBlockGrouping(3, false);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects1BlockActiveSpeech) {
+  CheckBlockGrouping(1, true);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects2BlocksActiveSpeech) {
+  CheckBlockGrouping(2, true);
+}
+
+TEST_F(AudioEncoderCngTest, EncodeCollects3BlocksActiveSpeech) {
+  CheckBlockGrouping(3, true);
+}
+
+TEST_F(AudioEncoderCngTest, EncodePassive) {
+  const size_t kBlocksPerFrame = 3;
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(kBlocksPerFrame));
+  auto config = MakeCngConfig();
+  const auto sid_frame_interval_ms = config.sid_frame_interval_ms;
+  const auto num_cng_coefficients = config.num_cng_coefficients;
+  CreateCng(std::move(config));
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillRepeatedly(Return(Vad::kPassive));
+  // Expect no calls at all to the speech encoder mock.
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).Times(0);
+  uint32_t expected_timestamp = timestamp_;
+  for (size_t i = 0; i < 100; ++i) {
+    Encode();
+    // Check if it was time to call the cng encoder. This is done once every
+    // `kBlocksPerFrame` calls.
+    if ((i + 1) % kBlocksPerFrame == 0) {
+      // Now check if a SID interval has elapsed.
+      if ((i % (sid_frame_interval_ms / 10)) < kBlocksPerFrame) {
+        // If so, verify that we got a CNG encoding.
+        EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+        EXPECT_FALSE(encoded_info_.speech);
+        EXPECT_EQ(static_cast<size_t>(num_cng_coefficients) + 1,
+                  encoded_info_.encoded_bytes);
+        EXPECT_EQ(expected_timestamp, encoded_info_.encoded_timestamp);
+      }
+      expected_timestamp += rtc::checked_cast<uint32_t>(
+          kBlocksPerFrame * num_audio_samples_10ms_);
+    } else {
+      // Otherwise, expect no output.
+      EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+    }
+  }
+}
+
+// Verifies that the correct action is taken for frames with both active and
+// passive speech.
+TEST_F(AudioEncoderCngTest, MixedActivePassive) {
+  CreateCng(MakeCngConfig());
+
+  // All of the frame is active speech.
+  ExpectEncodeCalls(6);
+  EXPECT_TRUE(CheckMixedActivePassive(Vad::kActive, Vad::kActive));
+  EXPECT_TRUE(encoded_info_.speech);
+
+  // First half of the frame is active speech.
+  ExpectEncodeCalls(6);
+  EXPECT_TRUE(CheckMixedActivePassive(Vad::kActive, Vad::kPassive));
+  EXPECT_TRUE(encoded_info_.speech);
+
+  // Second half of the frame is active speech.
+  ExpectEncodeCalls(6);
+  EXPECT_TRUE(CheckMixedActivePassive(Vad::kPassive, Vad::kActive));
+  EXPECT_TRUE(encoded_info_.speech);
+
+  // All of the frame is passive speech. Expect no calls to `mock_encoder_`.
+  EXPECT_FALSE(CheckMixedActivePassive(Vad::kPassive, Vad::kPassive));
+  EXPECT_FALSE(encoded_info_.speech);
+}
+
+// These tests verify that the audio is partitioned into larger blocks before
+// calling the VAD.
+// The parameters for CheckVadInputSize are:
+// CheckVadInputSize(frame_size, expected_first_block_size,
+//                   expected_second_block_size);
+TEST_F(AudioEncoderCngTest, VadInputSize10Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(10, 10, 0);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize20Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(20, 20, 0);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize30Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(30, 30, 0);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize40Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(40, 20, 20);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize50Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(50, 30, 20);
+}
+TEST_F(AudioEncoderCngTest, VadInputSize60Ms) {
+  CreateCng(MakeCngConfig());
+  CheckVadInputSize(60, 30, 30);
+}
+
+// Verifies that the correct payload type is set when CNG is encoded.
+TEST_F(AudioEncoderCngTest, VerifyCngPayloadType) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).Times(0);
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(1U));
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kPassive));
+  encoded_info_.payload_type = 0;
+  Encode();
+  EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+}
+
+// Verifies that a SID frame is encoded immediately as the signal changes from
+// active speech to passive.
+TEST_F(AudioEncoderCngTest, VerifySidFrameAfterSpeech) {
+  auto config = MakeCngConfig();
+  const auto num_cng_coefficients = config.num_cng_coefficients;
+  CreateCng(std::move(config));
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(1U));
+  // Start with encoding noise.
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .Times(2)
+      .WillRepeatedly(Return(Vad::kPassive));
+  Encode();
+  EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+  EXPECT_EQ(static_cast<size_t>(num_cng_coefficients) + 1,
+            encoded_info_.encoded_bytes);
+  // Encode again, and make sure we got no frame at all (since the SID frame
+  // period is 100 ms by default).
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+
+  // Now encode active speech.
+  encoded_info_.payload_type = 0;
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kActive));
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(
+          Invoke(MockAudioEncoder::FakeEncoding(kMockReturnEncodedBytes)));
+  Encode();
+  EXPECT_EQ(kMockReturnEncodedBytes, encoded_info_.encoded_bytes);
+
+  // Go back to noise again, and verify that a SID frame is emitted.
+  EXPECT_CALL(*mock_vad_, VoiceActivity(_, _, _))
+      .WillOnce(Return(Vad::kPassive));
+  Encode();
+  EXPECT_EQ(kCngPayloadType, encoded_info_.payload_type);
+  EXPECT_EQ(static_cast<size_t>(num_cng_coefficients) + 1,
+            encoded_info_.encoded_bytes);
+}
+
+// Resetting the CNG should reset both the VAD and the encoder.
+TEST_F(AudioEncoderCngTest, Reset) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, Reset()).Times(1);
+  EXPECT_CALL(*mock_vad_, Reset()).Times(1);
+  cng_->Reset();
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// This test fixture tests various error conditions that makes the
+// AudioEncoderCng die via CHECKs.
+class AudioEncoderCngDeathTest : public AudioEncoderCngTest {
+ protected:
+  AudioEncoderCngDeathTest() : AudioEncoderCngTest() {
+    EXPECT_CALL(*mock_vad_, Die()).Times(1);
+    delete mock_vad_;
+    mock_vad_ = nullptr;
+  }
+
+  // Override AudioEncoderCngTest::TearDown, since that one expects a call to
+  // the destructor of `mock_vad_`. In this case, that object is already
+  // deleted.
+  void TearDown() override { cng_.reset(); }
+
+  AudioEncoderCngConfig MakeCngConfig() {
+    // Don't provide a Vad mock object, since it would leak when the test dies.
+    auto config = AudioEncoderCngTest::MakeCngConfig();
+    config.vad = nullptr;
+    return config;
+  }
+
+  void TryWrongNumCoefficients(int num) {
+    RTC_EXPECT_DEATH(
+        [&] {
+          auto config = MakeCngConfig();
+          config.num_cng_coefficients = num;
+          CreateCng(std::move(config));
+        }(),
+        "Invalid configuration");
+  }
+};
+
+TEST_F(AudioEncoderCngDeathTest, WrongFrameSize) {
+  CreateCng(MakeCngConfig());
+  num_audio_samples_10ms_ *= 2;  // 20 ms frame.
+  RTC_EXPECT_DEATH(Encode(), "");
+  num_audio_samples_10ms_ = 0;  // Zero samples.
+  RTC_EXPECT_DEATH(Encode(), "");
+}
+
+TEST_F(AudioEncoderCngDeathTest, WrongNumCoefficientsA) {
+  TryWrongNumCoefficients(-1);
+}
+
+TEST_F(AudioEncoderCngDeathTest, WrongNumCoefficientsB) {
+  TryWrongNumCoefficients(0);
+}
+
+TEST_F(AudioEncoderCngDeathTest, WrongNumCoefficientsC) {
+  TryWrongNumCoefficients(13);
+}
+
+TEST_F(AudioEncoderCngDeathTest, NullSpeechEncoder) {
+  auto config = MakeCngConfig();
+  config.speech_encoder = nullptr;
+  RTC_EXPECT_DEATH(CreateCng(std::move(config)), "");
+}
+
+TEST_F(AudioEncoderCngDeathTest, StereoEncoder) {
+  EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(2));
+  RTC_EXPECT_DEATH(CreateCng(MakeCngConfig()), "Invalid configuration");
+}
+
+TEST_F(AudioEncoderCngDeathTest, StereoConfig) {
+  RTC_EXPECT_DEATH(
+      [&] {
+        auto config = MakeCngConfig();
+        config.num_channels = 2;
+        CreateCng(std::move(config));
+      }(),
+      "Invalid configuration");
+}
+
+TEST_F(AudioEncoderCngDeathTest, EncoderFrameSizeTooLarge) {
+  CreateCng(MakeCngConfig());
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillRepeatedly(Return(7U));
+  for (int i = 0; i < 6; ++i)
+    Encode();
+  RTC_EXPECT_DEATH(
+      Encode(), "Frame size cannot be larger than 60 ms when using VAD/CNG.");
+}
+
+#endif  // GTEST_HAS_DEATH_TEST
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/cng_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
new file mode 100644
index 0000000000..0e6ab79394
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
@@ -0,0 +1,252 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <memory>
+#include <string>
+
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+enum {
+  kSidShortIntervalUpdate = 1,
+  kSidNormalIntervalUpdate = 100,
+  kSidLongIntervalUpdate = 10000
+};
+
+enum : size_t {
+  kCNGNumParamsLow = 0,
+  kCNGNumParamsNormal = 8,
+  kCNGNumParamsHigh = WEBRTC_CNG_MAX_LPC_ORDER,
+  kCNGNumParamsTooHigh = WEBRTC_CNG_MAX_LPC_ORDER + 1
+};
+
+enum { kNoSid, kForceSid };
+
+class CngTest : public ::testing::Test {
+ protected:
+  virtual void SetUp();
+
+  void TestCngEncode(int sample_rate_hz, int quality);
+
+  int16_t speech_data_[640];  // Max size of CNG internal buffers.
+};
+
+class CngDeathTest : public CngTest {};
+
+void CngTest::SetUp() {
+  FILE* input_file;
+  const std::string file_name =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  input_file = fopen(file_name.c_str(), "rb");
+  ASSERT_TRUE(input_file != NULL);
+  ASSERT_EQ(640, static_cast<int32_t>(
+                     fread(speech_data_, sizeof(int16_t), 640, input_file)));
+  fclose(input_file);
+  input_file = NULL;
+}
+
+void CngTest::TestCngEncode(int sample_rate_hz, int quality) {
+  const size_t num_samples_10ms = rtc::CheckedDivExact(sample_rate_hz, 100);
+  rtc::Buffer sid_data;
+
+  ComfortNoiseEncoder cng_encoder(sample_rate_hz, kSidNormalIntervalUpdate,
+                                  quality);
+  EXPECT_EQ(0U, cng_encoder.Encode(rtc::ArrayView<const int16_t>(
+                                       speech_data_, num_samples_10ms),
+                                   kNoSid, &sid_data));
+  EXPECT_EQ(static_cast<size_t>(quality + 1),
+            cng_encoder.Encode(
+                rtc::ArrayView<const int16_t>(speech_data_, num_samples_10ms),
+                kForceSid, &sid_data));
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+// Create CNG encoder, init with faulty values, free CNG encoder.
+TEST_F(CngDeathTest, CngInitFail) {
+  // Call with too few parameters.
+  EXPECT_DEATH(
+      {
+        ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate, kCNGNumParamsLow);
+      },
+      "");
+  // Call with too many parameters.
+  EXPECT_DEATH(
+      {
+        ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate,
+                            kCNGNumParamsTooHigh);
+      },
+      "");
+}
+
+// Encode Cng with too long input vector.
+TEST_F(CngDeathTest, CngEncodeTooLong) {
+  rtc::Buffer sid_data;
+
+  // Create encoder.
+  ComfortNoiseEncoder cng_encoder(8000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  // Run encoder with too much data.
+  EXPECT_DEATH(
+      cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 641),
+                         kNoSid, &sid_data),
+      "");
+}
+#endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+TEST_F(CngTest, CngEncode8000) {
+  TestCngEncode(8000, kCNGNumParamsNormal);
+}
+
+TEST_F(CngTest, CngEncode16000) {
+  TestCngEncode(16000, kCNGNumParamsNormal);
+}
+
+TEST_F(CngTest, CngEncode32000) {
+  TestCngEncode(32000, kCNGNumParamsHigh);
+}
+
+TEST_F(CngTest, CngEncode48000) {
+  TestCngEncode(48000, kCNGNumParamsNormal);
+}
+
+TEST_F(CngTest, CngEncode64000) {
+  TestCngEncode(64000, kCNGNumParamsNormal);
+}
+
+// Update SID parameters, for both 9 and 16 parameters.
+TEST_F(CngTest, CngUpdateSid) {
+  rtc::Buffer sid_data;
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // Run normal Encode and UpdateSid.
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+  cng_decoder.UpdateSid(sid_data);
+
+  // Reinit with new length.
+  cng_encoder.Reset(16000, kSidNormalIntervalUpdate, kCNGNumParamsHigh);
+  cng_decoder.Reset();
+
+  // Expect 0 because of unstable parameters after switching length.
+  EXPECT_EQ(0U,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+  EXPECT_EQ(
+      kCNGNumParamsHigh + 1,
+      cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_ + 160, 160),
+                         kForceSid, &sid_data));
+  cng_decoder.UpdateSid(
+      rtc::ArrayView<const uint8_t>(sid_data.data(), kCNGNumParamsNormal + 1));
+}
+
+// Update SID parameters, with wrong parameters or without calling decode.
+TEST_F(CngTest, CngUpdateSidErroneous) {
+  rtc::Buffer sid_data;
+
+  // Encode.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+
+  // First run with valid parameters, then with too many CNG parameters.
+  // The function will operate correctly by only reading the maximum number of
+  // parameters, skipping the extra.
+  EXPECT_EQ(kCNGNumParamsNormal + 1, sid_data.size());
+  cng_decoder.UpdateSid(sid_data);
+
+  // Make sure the input buffer is large enough. Since Encode() appends data, we
+  // need to set the size manually only afterwards, or the buffer will be bigger
+  // than anticipated.
+  sid_data.SetSize(kCNGNumParamsTooHigh + 1);
+  cng_decoder.UpdateSid(sid_data);
+}
+
+// Test to generate cng data, by forcing SID. Both normal and faulty condition.
+TEST_F(CngTest, CngGenerate) {
+  rtc::Buffer sid_data;
+  int16_t out_data[640];
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // Normal Encode.
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kForceSid, &sid_data));
+
+  // Normal UpdateSid.
+  cng_decoder.UpdateSid(sid_data);
+
+  // Two normal Generate, one with new_period.
+  EXPECT_TRUE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 640), 1));
+  EXPECT_TRUE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 640), 0));
+
+  // Call Genereate with too much data.
+  EXPECT_FALSE(cng_decoder.Generate(rtc::ArrayView<int16_t>(out_data, 641), 0));
+}
+
+// Test automatic SID.
+TEST_F(CngTest, CngAutoSid) {
+  rtc::Buffer sid_data;
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidNormalIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // Normal Encode, 100 msec, where no SID data should be generated.
+  for (int i = 0; i < 10; i++) {
+    EXPECT_EQ(
+        0U, cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kNoSid, &sid_data));
+  }
+
+  // We have reached 100 msec, and SID data should be generated.
+  EXPECT_EQ(kCNGNumParamsNormal + 1,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kNoSid, &sid_data));
+}
+
+// Test automatic SID, with very short interval.
+TEST_F(CngTest, CngAutoSidShort) {
+  rtc::Buffer sid_data;
+
+  // Create and initialize encoder and decoder.
+  ComfortNoiseEncoder cng_encoder(16000, kSidShortIntervalUpdate,
+                                  kCNGNumParamsNormal);
+  ComfortNoiseDecoder cng_decoder;
+
+  // First call will never generate SID, unless forced to.
+  EXPECT_EQ(0U,
+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                               kNoSid, &sid_data));
+
+  // Normal Encode, 100 msec, SID data should be generated all the time.
+  for (int i = 0; i < 10; i++) {
+    EXPECT_EQ(
+        kCNGNumParamsNormal + 1,
+        cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
+                           kNoSid, &sid_data));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc b/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
new file mode 100644
index 0000000000..48f1b8c296
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
@@ -0,0 +1,436 @@
+/*
+ *  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 "modules/audio_coding/codecs/cng/webrtc_cng.h"
+
+#include <algorithm>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kCngMaxOutsizeOrder = 640;
+
+// TODO(ossu): Rename the left-over WebRtcCng according to style guide.
+void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a);
+
+const int32_t WebRtcCng_kDbov[94] = {
+    1081109975, 858756178, 682134279, 541838517, 430397633, 341876992,
+    271562548,  215709799, 171344384, 136103682, 108110997, 85875618,
+    68213428,   54183852,  43039763,  34187699,  27156255,  21570980,
+    17134438,   13610368,  10811100,  8587562,   6821343,   5418385,
+    4303976,    3418770,   2715625,   2157098,   1713444,   1361037,
+    1081110,    858756,    682134,    541839,    430398,    341877,
+    271563,     215710,    171344,    136104,    108111,    85876,
+    68213,      54184,     43040,     34188,     27156,     21571,
+    17134,      13610,     10811,     8588,      6821,      5418,
+    4304,       3419,      2716,      2157,      1713,      1361,
+    1081,       859,       682,       542,       430,       342,
+    272,        216,       171,       136,       108,       86,
+    68,         54,        43,        34,        27,        22,
+    17,         14,        11,        9,         7,         5,
+    4,          3,         3,         2,         2,         1,
+    1,          1,         1,         1};
+
+const int16_t WebRtcCng_kCorrWindow[WEBRTC_CNG_MAX_LPC_ORDER] = {
+    32702, 32636, 32570, 32505, 32439, 32374,
+    32309, 32244, 32179, 32114, 32049, 31985};
+
+}  // namespace
+
+ComfortNoiseDecoder::ComfortNoiseDecoder() {
+  /* Needed to get the right function pointers in SPLIB. */
+  Reset();
+}
+
+void ComfortNoiseDecoder::Reset() {
+  dec_seed_ = 7777; /* For debugging only. */
+  dec_target_energy_ = 0;
+  dec_used_energy_ = 0;
+  for (auto& c : dec_target_reflCoefs_)
+    c = 0;
+  for (auto& c : dec_used_reflCoefs_)
+    c = 0;
+  for (auto& c : dec_filtstate_)
+    c = 0;
+  for (auto& c : dec_filtstateLow_)
+    c = 0;
+  dec_order_ = 5;
+  dec_target_scale_factor_ = 0;
+  dec_used_scale_factor_ = 0;
+}
+
+void ComfortNoiseDecoder::UpdateSid(rtc::ArrayView<const uint8_t> sid) {
+  int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER];
+  int32_t targetEnergy;
+  size_t length = sid.size();
+  /* Throw away reflection coefficients of higher order than we can handle. */
+  if (length > (WEBRTC_CNG_MAX_LPC_ORDER + 1))
+    length = WEBRTC_CNG_MAX_LPC_ORDER + 1;
+
+  dec_order_ = static_cast<uint16_t>(length - 1);
+
+  uint8_t sid0 = std::min<uint8_t>(sid[0], 93);
+  targetEnergy = WebRtcCng_kDbov[sid0];
+  /* Take down target energy to 75%. */
+  targetEnergy = targetEnergy >> 1;
+  targetEnergy += targetEnergy >> 2;
+
+  dec_target_energy_ = targetEnergy;
+
+  /* Reconstruct coeffs with tweak for WebRtc implementation of RFC3389. */
+  if (dec_order_ == WEBRTC_CNG_MAX_LPC_ORDER) {
+    for (size_t i = 0; i < (dec_order_); i++) {
+      refCs[i] = sid[i + 1] << 8; /* Q7 to Q15*/
+      dec_target_reflCoefs_[i] = refCs[i];
+    }
+  } else {
+    for (size_t i = 0; i < (dec_order_); i++) {
+      refCs[i] = (sid[i + 1] - 127) * (1 << 8); /* Q7 to Q15. */
+      dec_target_reflCoefs_[i] = refCs[i];
+    }
+  }
+
+  for (size_t i = (dec_order_); i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
+    refCs[i] = 0;
+    dec_target_reflCoefs_[i] = refCs[i];
+  }
+}
+
+bool ComfortNoiseDecoder::Generate(rtc::ArrayView<int16_t> out_data,
+                                   bool new_period) {
+  int16_t excitation[kCngMaxOutsizeOrder];
+  int16_t low[kCngMaxOutsizeOrder];
+  int16_t lpPoly[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t ReflBetaStd = 26214;      /* 0.8 in q15. */
+  int16_t ReflBetaCompStd = 6553;   /* 0.2 in q15. */
+  int16_t ReflBetaNewP = 19661;     /* 0.6 in q15. */
+  int16_t ReflBetaCompNewP = 13107; /* 0.4 in q15. */
+  int16_t Beta, BetaC;              /* These are in Q15. */
+  int32_t targetEnergy;
+  int16_t En;
+  int16_t temp16;
+  const size_t num_samples = out_data.size();
+
+  if (num_samples > kCngMaxOutsizeOrder) {
+    return false;
+  }
+
+  if (new_period) {
+    dec_used_scale_factor_ = dec_target_scale_factor_;
+    Beta = ReflBetaNewP;
+    BetaC = ReflBetaCompNewP;
+  } else {
+    Beta = ReflBetaStd;
+    BetaC = ReflBetaCompStd;
+  }
+
+  /* Calculate new scale factor in Q13 */
+  dec_used_scale_factor_ = rtc::checked_cast<int16_t>(
+      WEBRTC_SPL_MUL_16_16_RSFT(dec_used_scale_factor_, Beta >> 2, 13) +
+      WEBRTC_SPL_MUL_16_16_RSFT(dec_target_scale_factor_, BetaC >> 2, 13));
+
+  dec_used_energy_ = dec_used_energy_ >> 1;
+  dec_used_energy_ += dec_target_energy_ >> 1;
+
+  /* Do the same for the reflection coeffs, albeit in Q15. */
+  for (size_t i = 0; i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
+    dec_used_reflCoefs_[i] =
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_used_reflCoefs_[i], Beta, 15);
+    dec_used_reflCoefs_[i] +=
+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_target_reflCoefs_[i], BetaC, 15);
+  }
+
+  /* Compute the polynomial coefficients. */
+  WebRtcCng_K2a16(dec_used_reflCoefs_, WEBRTC_CNG_MAX_LPC_ORDER, lpPoly);
+
+  targetEnergy = dec_used_energy_;
+
+  /* Calculate scaling factor based on filter energy. */
+  En = 8192; /* 1.0 in Q13. */
+  for (size_t i = 0; i < (WEBRTC_CNG_MAX_LPC_ORDER); i++) {
+    /* Floating point value for reference.
+       E *= 1.0 - (dec_used_reflCoefs_[i] / 32768.0) *
+       (dec_used_reflCoefs_[i] / 32768.0);
+     */
+
+    /* Same in fixed point. */
+    /* K(i).^2 in Q15. */
+    temp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_used_reflCoefs_[i],
+                                                dec_used_reflCoefs_[i], 15);
+    /* 1 - K(i).^2 in Q15. */
+    temp16 = 0x7fff - temp16;
+    En = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(En, temp16, 15);
+  }
+
+  /* float scaling= sqrt(E * dec_target_energy_ / (1 << 24)); */
+
+  /* Calculate sqrt(En * target_energy / excitation energy) */
+  targetEnergy = WebRtcSpl_Sqrt(dec_used_energy_);
+
+  En = (int16_t)WebRtcSpl_Sqrt(En) << 6;
+  En = (En * 3) >> 1; /* 1.5 estimates sqrt(2). */
+  dec_used_scale_factor_ = (int16_t)((En * targetEnergy) >> 12);
+
+  /* Generate excitation. */
+  /* Excitation energy per sample is 2.^24 - Q13 N(0,1). */
+  for (size_t i = 0; i < num_samples; i++) {
+    excitation[i] = WebRtcSpl_RandN(&dec_seed_) >> 1;
+  }
+
+  /* Scale to correct energy. */
+  WebRtcSpl_ScaleVector(excitation, excitation, dec_used_scale_factor_,
+                        num_samples, 13);
+
+  /* `lpPoly` - Coefficients in Q12.
+   * `excitation` - Speech samples.
+   * `nst->dec_filtstate` - State preservation.
+   * `out_data` - Filtered speech samples. */
+  WebRtcSpl_FilterAR(lpPoly, WEBRTC_CNG_MAX_LPC_ORDER + 1, excitation,
+                     num_samples, dec_filtstate_, WEBRTC_CNG_MAX_LPC_ORDER,
+                     dec_filtstateLow_, WEBRTC_CNG_MAX_LPC_ORDER,
+                     out_data.data(), low, num_samples);
+
+  return true;
+}
+
+ComfortNoiseEncoder::ComfortNoiseEncoder(int fs, int interval, int quality)
+    : enc_nrOfCoefs_(quality),
+      enc_sampfreq_(fs),
+      enc_interval_(interval),
+      enc_msSinceSid_(0),
+      enc_Energy_(0),
+      enc_reflCoefs_{0},
+      enc_corrVector_{0},
+      enc_seed_(7777) /* For debugging only. */ {
+  RTC_CHECK_GT(quality, 0);
+  RTC_CHECK_LE(quality, WEBRTC_CNG_MAX_LPC_ORDER);
+}
+
+void ComfortNoiseEncoder::Reset(int fs, int interval, int quality) {
+  RTC_CHECK_GT(quality, 0);
+  RTC_CHECK_LE(quality, WEBRTC_CNG_MAX_LPC_ORDER);
+  enc_nrOfCoefs_ = quality;
+  enc_sampfreq_ = fs;
+  enc_interval_ = interval;
+  enc_msSinceSid_ = 0;
+  enc_Energy_ = 0;
+  for (auto& c : enc_reflCoefs_)
+    c = 0;
+  for (auto& c : enc_corrVector_)
+    c = 0;
+  enc_seed_ = 7777; /* For debugging only. */
+}
+
+size_t ComfortNoiseEncoder::Encode(rtc::ArrayView<const int16_t> speech,
+                                   bool force_sid,
+                                   rtc::Buffer* output) {
+  int16_t arCoefs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int32_t corrVector[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t refCs[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t hanningW[kCngMaxOutsizeOrder];
+  int16_t ReflBeta = 19661;     /* 0.6 in q15. */
+  int16_t ReflBetaComp = 13107; /* 0.4 in q15. */
+  int32_t outEnergy;
+  int outShifts;
+  size_t i;
+  int stab;
+  int acorrScale;
+  size_t index;
+  size_t ind, factor;
+  int32_t* bptr;
+  int32_t blo, bhi;
+  int16_t negate;
+  const int16_t* aptr;
+  int16_t speechBuf[kCngMaxOutsizeOrder];
+
+  const size_t num_samples = speech.size();
+  RTC_CHECK_LE(num_samples, kCngMaxOutsizeOrder);
+
+  for (i = 0; i < num_samples; i++) {
+    speechBuf[i] = speech[i];
+  }
+
+  factor = num_samples;
+
+  /* Calculate energy and a coefficients. */
+  outEnergy = WebRtcSpl_Energy(speechBuf, num_samples, &outShifts);
+  while (outShifts > 0) {
+    /* We can only do 5 shifts without destroying accuracy in
+     * division factor. */
+    if (outShifts > 5) {
+      outEnergy <<= (outShifts - 5);
+      outShifts = 5;
+    } else {
+      factor /= 2;
+      outShifts--;
+    }
+  }
+  outEnergy = WebRtcSpl_DivW32W16(outEnergy, (int16_t)factor);
+
+  if (outEnergy > 1) {
+    /* Create Hanning Window. */
+    WebRtcSpl_GetHanningWindow(hanningW, num_samples / 2);
+    for (i = 0; i < (num_samples / 2); i++)
+      hanningW[num_samples - i - 1] = hanningW[i];
+
+    WebRtcSpl_ElementwiseVectorMult(speechBuf, hanningW, speechBuf, num_samples,
+                                    14);
+
+    WebRtcSpl_AutoCorrelation(speechBuf, num_samples, enc_nrOfCoefs_,
+                              corrVector, &acorrScale);
+
+    if (*corrVector == 0)
+      *corrVector = WEBRTC_SPL_WORD16_MAX;
+
+    /* Adds the bandwidth expansion. */
+    aptr = WebRtcCng_kCorrWindow;
+    bptr = corrVector;
+
+    /* (zzz) lpc16_1 = 17+1+820+2+2 = 842 (ordo2=700). */
+    for (ind = 0; ind < enc_nrOfCoefs_; ind++) {
+      /* The below code multiplies the 16 b corrWindow values (Q15) with
+       * the 32 b corrvector (Q0) and shifts the result down 15 steps. */
+      negate = *bptr < 0;
+      if (negate)
+        *bptr = -*bptr;
+
+      blo = (int32_t)*aptr * (*bptr & 0xffff);
+      bhi = ((blo >> 16) & 0xffff) +
+            ((int32_t)(*aptr++) * ((*bptr >> 16) & 0xffff));
+      blo = (blo & 0xffff) | ((bhi & 0xffff) << 16);
+
+      *bptr = (((bhi >> 16) & 0x7fff) << 17) | ((uint32_t)blo >> 15);
+      if (negate)
+        *bptr = -*bptr;
+      bptr++;
+    }
+    /* End of bandwidth expansion. */
+
+    stab = WebRtcSpl_LevinsonDurbin(corrVector, arCoefs, refCs, enc_nrOfCoefs_);
+
+    if (!stab) {
+      /* Disregard from this frame */
+      return 0;
+    }
+
+  } else {
+    for (i = 0; i < enc_nrOfCoefs_; i++)
+      refCs[i] = 0;
+  }
+
+  if (force_sid) {
+    /* Read instantaneous values instead of averaged. */
+    for (i = 0; i < enc_nrOfCoefs_; i++)
+      enc_reflCoefs_[i] = refCs[i];
+    enc_Energy_ = outEnergy;
+  } else {
+    /* Average history with new values. */
+    for (i = 0; i < enc_nrOfCoefs_; i++) {
+      enc_reflCoefs_[i] =
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(enc_reflCoefs_[i], ReflBeta, 15);
+      enc_reflCoefs_[i] +=
+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(refCs[i], ReflBetaComp, 15);
+    }
+    enc_Energy_ = (outEnergy >> 2) + (enc_Energy_ >> 1) + (enc_Energy_ >> 2);
+  }
+
+  if (enc_Energy_ < 1) {
+    enc_Energy_ = 1;
+  }
+
+  if ((enc_msSinceSid_ > (enc_interval_ - 1)) || force_sid) {
+    /* Search for best dbov value. */
+    index = 0;
+    for (i = 1; i < 93; i++) {
+      /* Always round downwards. */
+      if ((enc_Energy_ - WebRtcCng_kDbov[i]) > 0) {
+        index = i;
+        break;
+      }
+    }
+    if ((i == 93) && (index == 0))
+      index = 94;
+
+    const size_t output_coefs = enc_nrOfCoefs_ + 1;
+    output->AppendData(output_coefs, [&](rtc::ArrayView<uint8_t> output) {
+      output[0] = (uint8_t)index;
+
+      /* Quantize coefficients with tweak for WebRtc implementation of
+       * RFC3389. */
+      if (enc_nrOfCoefs_ == WEBRTC_CNG_MAX_LPC_ORDER) {
+        for (i = 0; i < enc_nrOfCoefs_; i++) {
+          /* Q15 to Q7 with rounding. */
+          output[i + 1] = ((enc_reflCoefs_[i] + 128) >> 8);
+        }
+      } else {
+        for (i = 0; i < enc_nrOfCoefs_; i++) {
+          /* Q15 to Q7 with rounding. */
+          output[i + 1] = (127 + ((enc_reflCoefs_[i] + 128) >> 8));
+        }
+      }
+
+      return output_coefs;
+    });
+
+    enc_msSinceSid_ =
+        static_cast<int16_t>((1000 * num_samples) / enc_sampfreq_);
+    return output_coefs;
+  } else {
+    enc_msSinceSid_ +=
+        static_cast<int16_t>((1000 * num_samples) / enc_sampfreq_);
+    return 0;
+  }
+}
+
+namespace {
+/* Values in `k` are Q15, and `a` Q12. */
+void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a) {
+  int16_t any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
+  int16_t* aptr;
+  int16_t* aptr2;
+  int16_t* anyptr;
+  const int16_t* kptr;
+  int m, i;
+
+  kptr = k;
+  *a = 4096; /* i.e., (Word16_MAX >> 3) + 1 */
+  *any = *a;
+  a[1] = (*k + 4) >> 3;
+  for (m = 1; m < useOrder; m++) {
+    kptr++;
+    aptr = a;
+    aptr++;
+    aptr2 = &a[m];
+    anyptr = any;
+    anyptr++;
+
+    any[m + 1] = (*kptr + 4) >> 3;
+    for (i = 0; i < m; i++) {
+      *anyptr++ =
+          (*aptr++) +
+          (int16_t)((((int32_t)(*aptr2--) * (int32_t)*kptr) + 16384) >> 15);
+    }
+
+    aptr = a;
+    anyptr = any;
+    for (i = 0; i < (m + 2); i++) {
+      *aptr++ = *anyptr++;
+    }
+  }
+}
+
+}  // namespace
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.h b/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
new file mode 100644
index 0000000000..7afd243f81
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
@@ -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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+#define MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+
+#include <stdint.h>
+
+#include <cstddef>
+
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+
+#define WEBRTC_CNG_MAX_LPC_ORDER 12
+
+namespace webrtc {
+
+class ComfortNoiseDecoder {
+ public:
+  ComfortNoiseDecoder();
+  ~ComfortNoiseDecoder() = default;
+
+  ComfortNoiseDecoder(const ComfortNoiseDecoder&) = delete;
+  ComfortNoiseDecoder& operator=(const ComfortNoiseDecoder&) = delete;
+
+  void Reset();
+
+  // Updates the CN state when a new SID packet arrives.
+  // `sid` is a view of the SID packet without the headers.
+  void UpdateSid(rtc::ArrayView<const uint8_t> sid);
+
+  // Generates comfort noise.
+  // `out_data` will be filled with samples - its size determines the number of
+  // samples generated. When `new_period` is true, CNG history will be reset
+  // before any audio is generated.  Returns `false` if outData is too large -
+  // currently 640 bytes (equalling 10ms at 64kHz).
+  // TODO(ossu): Specify better limits for the size of out_data. Either let it
+  //             be unbounded or limit to 10ms in the current sample rate.
+  bool Generate(rtc::ArrayView<int16_t> out_data, bool new_period);
+
+ private:
+  uint32_t dec_seed_;
+  int32_t dec_target_energy_;
+  int32_t dec_used_energy_;
+  int16_t dec_target_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_used_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_filtstate_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int16_t dec_filtstateLow_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  uint16_t dec_order_;
+  int16_t dec_target_scale_factor_; /* Q29 */
+  int16_t dec_used_scale_factor_;   /* Q29 */
+};
+
+class ComfortNoiseEncoder {
+ public:
+  // Creates a comfort noise encoder.
+  // `fs` selects sample rate: 8000 for narrowband or 16000 for wideband.
+  // `interval` sets the interval at which to generate SID data (in ms).
+  // `quality` selects the number of refl. coeffs. Maximum allowed is 12.
+  ComfortNoiseEncoder(int fs, int interval, int quality);
+  ~ComfortNoiseEncoder() = default;
+
+  ComfortNoiseEncoder(const ComfortNoiseEncoder&) = delete;
+  ComfortNoiseEncoder& operator=(const ComfortNoiseEncoder&) = delete;
+
+  // Resets the comfort noise encoder to its initial state.
+  // Parameters are set as during construction.
+  void Reset(int fs, int interval, int quality);
+
+  // Analyzes background noise from `speech` and appends coefficients to
+  // `output`.  Returns the number of coefficients generated.  If `force_sid` is
+  // true, a SID frame is forced and the internal sid interval counter is reset.
+  // Will fail if the input size is too large (> 640 samples, see
+  // ComfortNoiseDecoder::Generate).
+  size_t Encode(rtc::ArrayView<const int16_t> speech,
+                bool force_sid,
+                rtc::Buffer* output);
+
+ private:
+  size_t enc_nrOfCoefs_;
+  int enc_sampfreq_;
+  int16_t enc_interval_;
+  int16_t enc_msSinceSid_;
+  int32_t enc_Energy_;
+  int16_t enc_reflCoefs_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  int32_t enc_corrVector_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
+  uint32_t enc_seed_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
new file mode 100644
index 0000000000..46ac671b30
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+
+namespace webrtc {
+
+void AudioDecoderPcmU::Reset() {}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmU::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, 8 * num_channels_, 8);
+}
+
+int AudioDecoderPcmU::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderPcmU::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmU::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  // Adjust the encoded length down to ensure the same number of samples in each
+  // channel.
+  const size_t encoded_len_adjusted =
+      PacketDuration(encoded, encoded_len) *
+      Channels();         // 1 byte per sample per channel
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret =
+      WebRtcG711_DecodeU(encoded, encoded_len_adjusted, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+void AudioDecoderPcmA::Reset() {}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, 8 * num_channels_, 8);
+}
+
+int AudioDecoderPcmA::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderPcmA::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcmA::DecodeInternal(const uint8_t* encoded,
+                                     size_t encoded_len,
+                                     int sample_rate_hz,
+                                     int16_t* decoded,
+                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  // Adjust the encoded length down to ensure the same number of samples in each
+  // channel.
+  const size_t encoded_len_adjusted =
+      PacketDuration(encoded, encoded_len) *
+      Channels();         // 1 byte per sample per channel
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret =
+      WebRtcG711_DecodeA(encoded, encoded_len_adjusted, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
+                                     size_t encoded_len) const {
+  // One encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / Channels());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
new file mode 100644
index 0000000000..3fa42cba30
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class AudioDecoderPcmU final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
+    RTC_DCHECK_GE(num_channels, 1);
+  }
+
+  AudioDecoderPcmU(const AudioDecoderPcmU&) = delete;
+  AudioDecoderPcmU& operator=(const AudioDecoderPcmU&) = delete;
+
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const size_t num_channels_;
+};
+
+class AudioDecoderPcmA final : public AudioDecoder {
+ public:
+  explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
+    RTC_DCHECK_GE(num_channels, 1);
+  }
+
+  AudioDecoderPcmA(const AudioDecoderPcmA&) = delete;
+  AudioDecoderPcmA& operator=(const AudioDecoderPcmA&) = delete;
+
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const size_t num_channels_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
new file mode 100644
index 0000000000..65e2da479d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+bool AudioEncoderPcm::Config::IsOk() const {
+  return (frame_size_ms % 10 == 0) && (num_channels >= 1);
+}
+
+AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
+    : sample_rate_hz_(sample_rate_hz),
+      num_channels_(config.num_channels),
+      payload_type_(config.payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      full_frame_samples_(config.num_channels * config.frame_size_ms *
+                          sample_rate_hz / 1000),
+      first_timestamp_in_buffer_(0) {
+  RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
+  RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
+      << "Frame size must be an integer multiple of 10 ms.";
+  speech_buffer_.reserve(full_frame_samples_);
+}
+
+AudioEncoderPcm::~AudioEncoderPcm() = default;
+
+int AudioEncoderPcm::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
+size_t AudioEncoderPcm::NumChannels() const {
+  return num_channels_;
+}
+
+size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderPcm::GetTargetBitrate() const {
+  return static_cast<int>(8 * BytesPerSample() * SampleRateHz() *
+                          NumChannels());
+}
+
+AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (speech_buffer_.empty()) {
+    first_timestamp_in_buffer_ = rtp_timestamp;
+  }
+  speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
+  if (speech_buffer_.size() < full_frame_samples_) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
+  EncodedInfo info;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoded_bytes = encoded->AppendData(
+      full_frame_samples_ * BytesPerSample(),
+      [&](rtc::ArrayView<uint8_t> encoded) {
+        return EncodeCall(&speech_buffer_[0], full_frame_samples_,
+                          encoded.data());
+      });
+  speech_buffer_.clear();
+  info.encoder_type = GetCodecType();
+  return info;
+}
+
+void AudioEncoderPcm::Reset() {
+  speech_buffer_.clear();
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderPcm::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeA(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmA::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoder::CodecType AudioEncoderPcmA::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmA;
+}
+
+size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeU(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcmU::BytesPerSample() const {
+  return 1;
+}
+
+AudioEncoder::CodecType AudioEncoderPcmU::GetCodecType() const {
+  return AudioEncoder::CodecType::kPcmU;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
new file mode 100644
index 0000000000..d50be4b457
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+class AudioEncoderPcm : public AudioEncoder {
+ public:
+  struct Config {
+   public:
+    bool IsOk() const;
+
+    int frame_size_ms;
+    size_t num_channels;
+    int payload_type;
+
+   protected:
+    explicit Config(int pt)
+        : frame_size_ms(20), num_channels(1), payload_type(pt) {}
+  };
+
+  ~AudioEncoderPcm() override;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  AudioEncoderPcm(const Config& config, int sample_rate_hz);
+
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+  virtual size_t EncodeCall(const int16_t* audio,
+                            size_t input_len,
+                            uint8_t* encoded) = 0;
+
+  virtual size_t BytesPerSample() const = 0;
+
+  // Used to set EncodedInfoLeaf::encoder_type in
+  // AudioEncoderPcm::EncodeImpl
+  virtual AudioEncoder::CodecType GetCodecType() const = 0;
+
+ private:
+  const int sample_rate_hz_;
+  const size_t num_channels_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  const size_t full_frame_samples_;
+  std::vector<int16_t> speech_buffer_;
+  uint32_t first_timestamp_in_buffer_;
+};
+
+class AudioEncoderPcmA final : public AudioEncoderPcm {
+ public:
+  struct Config : public AudioEncoderPcm::Config {
+    Config() : AudioEncoderPcm::Config(8) {}
+  };
+
+  explicit AudioEncoderPcmA(const Config& config)
+      : AudioEncoderPcm(config, kSampleRateHz) {}
+
+  AudioEncoderPcmA(const AudioEncoderPcmA&) = delete;
+  AudioEncoderPcmA& operator=(const AudioEncoderPcmA&) = delete;
+
+ protected:
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+  AudioEncoder::CodecType GetCodecType() const override;
+
+ private:
+  static const int kSampleRateHz = 8000;
+};
+
+class AudioEncoderPcmU final : public AudioEncoderPcm {
+ public:
+  struct Config : public AudioEncoderPcm::Config {
+    Config() : AudioEncoderPcm::Config(0) {}
+  };
+
+  explicit AudioEncoderPcmU(const Config& config)
+      : AudioEncoderPcm(config, kSampleRateHz) {}
+
+  AudioEncoderPcmU(const AudioEncoderPcmU&) = delete;
+  AudioEncoderPcmU& operator=(const AudioEncoderPcmU&) = delete;
+
+ protected:
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+  AudioEncoder::CodecType GetCodecType() const override;
+
+ private:
+  static const int kSampleRateHz = 8000;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c
new file mode 100644
index 0000000000..5fe1692ccb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.c
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string.h>
+
+#include "modules/third_party/g711/g711.h"
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+
+size_t WebRtcG711_EncodeA(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    encoded[n] = linear_to_alaw(speechIn[n]);
+  return len;
+}
+
+size_t WebRtcG711_EncodeU(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    encoded[n] = linear_to_ulaw(speechIn[n]);
+  return len;
+}
+
+size_t WebRtcG711_DecodeA(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    decoded[n] = alaw_to_linear(encoded[n]);
+  *speechType = 1;
+  return len;
+}
+
+size_t WebRtcG711_DecodeU(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType) {
+  size_t n;
+  for (n = 0; n < len; n++)
+    decoded[n] = ulaw_to_linear(encoded[n]);
+  *speechType = 1;
+  return len;
+}
+
+int16_t WebRtcG711_Version(char* version, int16_t lenBytes) {
+  strncpy(version, "2.0.0", lenBytes);
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h
new file mode 100644
index 0000000000..c92e6cc1c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/g711_interface.h
@@ -0,0 +1,136 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+// Comfort noise constants
+#define G711_WEBRTC_SPEECH 1
+#define G711_WEBRTC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcG711_EncodeA(...)
+ *
+ * This function encodes a G711 A-law frame and inserts it into a packet.
+ * Input speech length has be of any length.
+ *
+ * Input:
+ *      - speechIn           : Input speech vector
+ *      - len                : Samples in speechIn
+ *
+ * Output:
+ *      - encoded            : The encoded data vector
+ *
+ * Return value              : Length (in bytes) of coded data.
+ *                             Always equal to len input parameter.
+ */
+
+size_t WebRtcG711_EncodeA(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG711_EncodeU(...)
+ *
+ * This function encodes a G711 U-law frame and inserts it into a packet.
+ * Input speech length has be of any length.
+ *
+ * Input:
+ *      - speechIn           : Input speech vector
+ *      - len                : Samples in speechIn
+ *
+ * Output:
+ *      - encoded            : The encoded data vector
+ *
+ * Return value              : Length (in bytes) of coded data.
+ *                             Always equal to len input parameter.
+ */
+
+size_t WebRtcG711_EncodeU(const int16_t* speechIn,
+                          size_t len,
+                          uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG711_DecodeA(...)
+ *
+ * This function decodes a packet G711 A-law frame.
+ *
+ * Input:
+ *      - encoded            : Encoded data
+ *      - len                : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - speechType         : 1 normal, 2 CNG (for G711 it should
+ *                             always return 1 since G711 does not have a
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples in decoded vector
+ *                             -1 - Error
+ */
+
+size_t WebRtcG711_DecodeA(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType);
+
+/****************************************************************************
+ * WebRtcG711_DecodeU(...)
+ *
+ * This function decodes a packet G711 U-law frame.
+ *
+ * Input:
+ *      - encoded            : Encoded data
+ *      - len                : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - speechType         : 1 normal, 2 CNG (for G711 it should
+ *                             always return 1 since G711 does not have a
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples in decoded vector
+ *                             -1 - Error
+ */
+
+size_t WebRtcG711_DecodeU(const uint8_t* encoded,
+                          size_t len,
+                          int16_t* decoded,
+                          int16_t* speechType);
+
+/**********************************************************************
+ * WebRtcG711_Version(...)
+ *
+ * This function gives the version string of the G.711 codec.
+ *
+ * Input:
+ *      - lenBytes:     the size of Allocated space (in Bytes) where
+ *                      the version number is written to (in string format).
+ *
+ * Output:
+ *      - version:      Pointer to a buffer where the version number is
+ *                      written to.
+ *
+ */
+
+int16_t WebRtcG711_Version(char* version, int16_t lenBytes);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc
new file mode 100644
index 0000000000..f3a42f5d79
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/test/testG711.cc
@@ -0,0 +1,168 @@
+/*
+ *  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.
+ */
+
+/*
+ * testG711.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+/* include API */
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G711 1000
+
+/* function for reading audio data from PCM file */
+bool readframe(int16_t* data, FILE* inp, size_t length) {
+  size_t rlen = fread(data, sizeof(int16_t), length, inp);
+  if (rlen >= length)
+    return false;
+  memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
+  return true;
+}
+
+int main(int argc, char* argv[]) {
+  char inname[80], outname[40], bitname[40];
+  FILE* inp;
+  FILE* outp;
+  FILE* bitp = NULL;
+  int framecnt;
+  bool endfile;
+
+  size_t framelength = 80;
+
+  /* Runtime statistics */
+  double starttime;
+  double runtime;
+  double length_file;
+
+  size_t stream_len = 0;
+  int16_t shortdata[480];
+  int16_t decoded[480];
+  uint8_t streamdata[1000];
+  int16_t speechType[1];
+  char law[2];
+  char versionNumber[40];
+
+  /* handling wrong input arguments in the command line */
+  if ((argc != 5) && (argc != 6)) {
+    printf("\n\nWrong number of arguments or flag values.\n\n");
+
+    printf("\n");
+    printf("\nG.711 test application\n\n");
+    printf("Usage:\n\n");
+    printf("./testG711.exe framelength law infile outfile \n\n");
+    printf("framelength: Framelength in samples.\n");
+    printf("law        : Coding law, A och u.\n");
+    printf("infile     : Normal speech input file\n");
+    printf("outfile    : Speech output file\n\n");
+    printf("outbits    : Output bitstream file [optional]\n\n");
+    exit(0);
+  }
+
+  /* Get version and print */
+  WebRtcG711_Version(versionNumber, 40);
+
+  printf("-----------------------------------\n");
+  printf("G.711 version: %s\n\n", versionNumber);
+  /* Get frame length */
+  int framelength_int = atoi(argv[1]);
+  if (framelength_int < 0) {
+    printf("  G.722: Invalid framelength %d.\n", framelength_int);
+    exit(1);
+  }
+  framelength = static_cast<size_t>(framelength_int);
+
+  /* Get compression law */
+  strcpy(law, argv[2]);
+
+  /* Get Input and Output files */
+  sscanf(argv[3], "%s", inname);
+  sscanf(argv[4], "%s", outname);
+  if (argc == 6) {
+    sscanf(argv[5], "%s", bitname);
+    if ((bitp = fopen(bitname, "wb")) == NULL) {
+      printf("  G.711: Cannot read file %s.\n", bitname);
+      exit(1);
+    }
+  }
+
+  if ((inp = fopen(inname, "rb")) == NULL) {
+    printf("  G.711: Cannot read file %s.\n", inname);
+    exit(1);
+  }
+  if ((outp = fopen(outname, "wb")) == NULL) {
+    printf("  G.711: Cannot write file %s.\n", outname);
+    exit(1);
+  }
+  printf("\nInput:  %s\nOutput: %s\n", inname, outname);
+  if (argc == 6) {
+    printf("\nBitfile:  %s\n", bitname);
+  }
+
+  starttime = clock() / (double)CLOCKS_PER_SEC_G711; /* Runtime statistics */
+
+  /* Initialize encoder and decoder */
+  framecnt = 0;
+  endfile = false;
+  while (!endfile) {
+    framecnt++;
+    /* Read speech block */
+    endfile = readframe(shortdata, inp, framelength);
+
+    /* G.711 encoding */
+    if (!strcmp(law, "A")) {
+      /* A-law encoding */
+      stream_len = WebRtcG711_EncodeA(shortdata, framelength, streamdata);
+      if (argc == 6) {
+        /* Write bits to file */
+        if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
+            stream_len) {
+          return -1;
+        }
+      }
+      WebRtcG711_DecodeA(streamdata, stream_len, decoded, speechType);
+    } else if (!strcmp(law, "u")) {
+      /* u-law encoding */
+      stream_len = WebRtcG711_EncodeU(shortdata, framelength, streamdata);
+      if (argc == 6) {
+        /* Write bits to file */
+        if (fwrite(streamdata, sizeof(unsigned char), stream_len, bitp) !=
+            stream_len) {
+          return -1;
+        }
+      }
+      WebRtcG711_DecodeU(streamdata, stream_len, decoded, speechType);
+    } else {
+      printf("Wrong law mode\n");
+      exit(1);
+    }
+    /* Write coded speech to file */
+    if (fwrite(decoded, sizeof(short), framelength, outp) != framelength) {
+      return -1;
+    }
+  }
+
+  runtime = (double)(clock() / (double)CLOCKS_PER_SEC_G711 - starttime);
+  length_file = ((double)framecnt * (double)framelength / 8000);
+  printf("\n\nLength of speech file: %.1f s\n", length_file);
+  printf("Time to run G.711:      %.2f s (%.2f %% of realtime)\n\n", runtime,
+         (100 * runtime / length_file));
+  printf("---------------------END----------------------\n");
+
+  fclose(inp);
+  fclose(outp);
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
new file mode 100644
index 0000000000..1ecc9bc3d1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
@@ -0,0 +1,178 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+
+#include <string.h>
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderG722Impl::AudioDecoderG722Impl() {
+  WebRtcG722_CreateDecoder(&dec_state_);
+  WebRtcG722_DecoderInit(dec_state_);
+}
+
+AudioDecoderG722Impl::~AudioDecoderG722Impl() {
+  WebRtcG722_FreeDecoder(dec_state_);
+}
+
+bool AudioDecoderG722Impl::HasDecodePlc() const {
+  return false;
+}
+
+int AudioDecoderG722Impl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  int16_t temp_type = 1;  // Default is speech.
+  size_t ret =
+      WebRtcG722_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return static_cast<int>(ret);
+}
+
+void AudioDecoderG722Impl::Reset() {
+  WebRtcG722_DecoderInit(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722Impl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
+                                                 timestamp, 8, 16);
+}
+
+int AudioDecoderG722Impl::PacketDuration(const uint8_t* encoded,
+                                         size_t encoded_len) const {
+  // 1/2 encoded byte per sample per channel.
+  return static_cast<int>(2 * encoded_len / Channels());
+}
+
+int AudioDecoderG722Impl::SampleRateHz() const {
+  return 16000;
+}
+
+size_t AudioDecoderG722Impl::Channels() const {
+  return 1;
+}
+
+AudioDecoderG722StereoImpl::AudioDecoderG722StereoImpl() {
+  WebRtcG722_CreateDecoder(&dec_state_left_);
+  WebRtcG722_CreateDecoder(&dec_state_right_);
+  WebRtcG722_DecoderInit(dec_state_left_);
+  WebRtcG722_DecoderInit(dec_state_right_);
+}
+
+AudioDecoderG722StereoImpl::~AudioDecoderG722StereoImpl() {
+  WebRtcG722_FreeDecoder(dec_state_left_);
+  WebRtcG722_FreeDecoder(dec_state_right_);
+}
+
+int AudioDecoderG722StereoImpl::DecodeInternal(const uint8_t* encoded,
+                                               size_t encoded_len,
+                                               int sample_rate_hz,
+                                               int16_t* decoded,
+                                               SpeechType* speech_type) {
+  RTC_DCHECK_EQ(SampleRateHz(), sample_rate_hz);
+  // Adjust the encoded length down to ensure the same number of samples in each
+  // channel.
+  const size_t encoded_len_adjusted = PacketDuration(encoded, encoded_len) *
+                                      Channels() /
+                                      2;  // 1/2 byte per sample per channel
+  int16_t temp_type = 1;  // Default is speech.
+  // De-interleave the bit-stream into two separate payloads.
+  uint8_t* encoded_deinterleaved = new uint8_t[encoded_len_adjusted];
+  SplitStereoPacket(encoded, encoded_len_adjusted, encoded_deinterleaved);
+  // Decode left and right.
+  size_t decoded_len =
+      WebRtcG722_Decode(dec_state_left_, encoded_deinterleaved,
+                        encoded_len_adjusted / 2, decoded, &temp_type);
+  size_t ret = WebRtcG722_Decode(
+      dec_state_right_, &encoded_deinterleaved[encoded_len_adjusted / 2],
+      encoded_len_adjusted / 2, &decoded[decoded_len], &temp_type);
+  if (ret == decoded_len) {
+    ret += decoded_len;  // Return total number of samples.
+    // Interleave output.
+    for (size_t k = ret / 2; k < ret; k++) {
+      int16_t temp = decoded[k];
+      memmove(&decoded[2 * k - ret + 2], &decoded[2 * k - ret + 1],
+              (ret - k - 1) * sizeof(int16_t));
+      decoded[2 * k - ret + 1] = temp;
+    }
+  }
+  *speech_type = ConvertSpeechType(temp_type);
+  delete[] encoded_deinterleaved;
+  return static_cast<int>(ret);
+}
+
+int AudioDecoderG722StereoImpl::PacketDuration(const uint8_t* encoded,
+                                               size_t encoded_len) const {
+  // 1/2 encoded byte per sample per channel. Make sure the length represents
+  // an equal number of bytes per channel. Otherwise, we cannot de-interleave
+  // the encoded data later.
+  return static_cast<int>(2 * (encoded_len / Channels()));
+}
+
+int AudioDecoderG722StereoImpl::SampleRateHz() const {
+  return 16000;
+}
+
+size_t AudioDecoderG722StereoImpl::Channels() const {
+  return 2;
+}
+
+void AudioDecoderG722StereoImpl::Reset() {
+  WebRtcG722_DecoderInit(dec_state_left_);
+  WebRtcG722_DecoderInit(dec_state_right_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderG722StereoImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  return LegacyEncodedAudioFrame::SplitBySamples(this, std::move(payload),
+                                                 timestamp, 2 * 8, 16);
+}
+
+// Split the stereo packet and place left and right channel after each other
+// in the output array.
+void AudioDecoderG722StereoImpl::SplitStereoPacket(
+    const uint8_t* encoded,
+    size_t encoded_len,
+    uint8_t* encoded_deinterleaved) {
+  // Regroup the 4 bits/sample so |l1 l2| |r1 r2| |l3 l4| |r3 r4| ...,
+  // where "lx" is 4 bits representing left sample number x, and "rx" right
+  // sample. Two samples fit in one byte, represented with |...|.
+  for (size_t i = 0; i + 1 < encoded_len; i += 2) {
+    uint8_t right_byte = ((encoded[i] & 0x0F) << 4) + (encoded[i + 1] & 0x0F);
+    encoded_deinterleaved[i] = (encoded[i] & 0xF0) + (encoded[i + 1] >> 4);
+    encoded_deinterleaved[i + 1] = right_byte;
+  }
+
+  // Move one byte representing right channel each loop, and place it at the
+  // end of the bytestream vector. After looping the data is reordered to:
+  // |l1 l2| |l3 l4| ... |l(N-1) lN| |r1 r2| |r3 r4| ... |r(N-1) r(N)|,
+  // where N is the total number of samples.
+  for (size_t i = 0; i < encoded_len / 2; i++) {
+    uint8_t right_byte = encoded_deinterleaved[i + 1];
+    memmove(&encoded_deinterleaved[i + 1], &encoded_deinterleaved[i + 2],
+            encoded_len - i - 2);
+    encoded_deinterleaved[encoded_len - 1] = right_byte;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
new file mode 100644
index 0000000000..5872fad5de
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+
+#include "api/audio_codecs/audio_decoder.h"
+
+typedef struct WebRtcG722DecInst G722DecInst;
+
+namespace webrtc {
+
+class AudioDecoderG722Impl final : public AudioDecoder {
+ public:
+  AudioDecoderG722Impl();
+  ~AudioDecoderG722Impl() override;
+
+  AudioDecoderG722Impl(const AudioDecoderG722Impl&) = delete;
+  AudioDecoderG722Impl& operator=(const AudioDecoderG722Impl&) = delete;
+
+  bool HasDecodePlc() const override;
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  G722DecInst* dec_state_;
+};
+
+class AudioDecoderG722StereoImpl final : public AudioDecoder {
+ public:
+  AudioDecoderG722StereoImpl();
+  ~AudioDecoderG722StereoImpl() override;
+
+  AudioDecoderG722StereoImpl(const AudioDecoderG722StereoImpl&) = delete;
+  AudioDecoderG722StereoImpl& operator=(const AudioDecoderG722StereoImpl&) =
+      delete;
+
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int SampleRateHz() const override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  // Splits the stereo-interleaved payload in `encoded` into separate payloads
+  // for left and right channels. The separated payloads are written to
+  // `encoded_deinterleaved`, which must hold at least `encoded_len` samples.
+  // The left channel starts at offset 0, while the right channel starts at
+  // offset encoded_len / 2 into `encoded_deinterleaved`.
+  void SplitStereoPacket(const uint8_t* encoded,
+                         size_t encoded_len,
+                         uint8_t* encoded_deinterleaved);
+
+  G722DecInst* dec_state_left_;
+  G722DecInst* dec_state_right_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
new file mode 100644
index 0000000000..b7d34ba581
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const size_t kSampleRateHz = 16000;
+
+}  // namespace
+
+AudioEncoderG722Impl::AudioEncoderG722Impl(const AudioEncoderG722Config& config,
+                                           int payload_type)
+    : num_channels_(config.num_channels),
+      payload_type_(payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      num_10ms_frames_buffered_(0),
+      first_timestamp_in_buffer_(0),
+      encoders_(new EncoderState[num_channels_]),
+      interleave_buffer_(2 * num_channels_) {
+  RTC_CHECK(config.IsOk());
+  const size_t samples_per_channel =
+      kSampleRateHz / 100 * num_10ms_frames_per_packet_;
+  for (size_t i = 0; i < num_channels_; ++i) {
+    encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
+    encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
+  }
+  Reset();
+}
+
+AudioEncoderG722Impl::~AudioEncoderG722Impl() = default;
+
+int AudioEncoderG722Impl::SampleRateHz() const {
+  return kSampleRateHz;
+}
+
+size_t AudioEncoderG722Impl::NumChannels() const {
+  return num_channels_;
+}
+
+int AudioEncoderG722Impl::RtpTimestampRateHz() const {
+  // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
+  // codec.
+  return kSampleRateHz / 2;
+}
+
+size_t AudioEncoderG722Impl::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderG722Impl::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderG722Impl::GetTargetBitrate() const {
+  // 4 bits/sample, 16000 samples/s/channel.
+  return static_cast<int>(64000 * NumChannels());
+}
+
+void AudioEncoderG722Impl::Reset() {
+  num_10ms_frames_buffered_ = 0;
+  for (size_t i = 0; i < num_channels_; ++i)
+    RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderG722Impl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+AudioEncoder::EncodedInfo AudioEncoderG722Impl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (num_10ms_frames_buffered_ == 0)
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  // Deinterleave samples and save them in each channel's buffer.
+  const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
+  for (size_t i = 0; i < kSampleRateHz / 100; ++i)
+    for (size_t j = 0; j < num_channels_; ++j)
+      encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
+
+  // If we don't yet have enough samples for a packet, we're done for now.
+  if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
+    return EncodedInfo();
+  }
+
+  // Encode each channel separately.
+  RTC_CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  num_10ms_frames_buffered_ = 0;
+  const size_t samples_per_channel = SamplesPerChannel();
+  for (size_t i = 0; i < num_channels_; ++i) {
+    const size_t bytes_encoded = WebRtcG722_Encode(
+        encoders_[i].encoder, encoders_[i].speech_buffer.get(),
+        samples_per_channel, encoders_[i].encoded_buffer.data());
+    RTC_CHECK_EQ(bytes_encoded, samples_per_channel / 2);
+  }
+
+  const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_;
+  EncodedInfo info;
+  info.encoded_bytes = encoded->AppendData(
+      bytes_to_encode, [&](rtc::ArrayView<uint8_t> encoded) {
+        // Interleave the encoded bytes of the different channels. Each separate
+        // channel and the interleaved stream encodes two samples per byte, most
+        // significant half first.
+        for (size_t i = 0; i < samples_per_channel / 2; ++i) {
+          for (size_t j = 0; j < num_channels_; ++j) {
+            uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
+            interleave_buffer_.data()[j] = two_samples >> 4;
+            interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
+          }
+          for (size_t j = 0; j < num_channels_; ++j)
+            encoded[i * num_channels_ + j] =
+                interleave_buffer_.data()[2 * j] << 4 |
+                interleave_buffer_.data()[2 * j + 1];
+        }
+
+        return bytes_to_encode;
+      });
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kG722;
+  return info;
+}
+
+AudioEncoderG722Impl::EncoderState::EncoderState() {
+  RTC_CHECK_EQ(0, WebRtcG722_CreateEncoder(&encoder));
+}
+
+AudioEncoderG722Impl::EncoderState::~EncoderState() {
+  RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
+}
+
+size_t AudioEncoderG722Impl::SamplesPerChannel() const {
+  return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
new file mode 100644
index 0000000000..a932aa8b7d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioEncoderG722Impl final : public AudioEncoder {
+ public:
+  AudioEncoderG722Impl(const AudioEncoderG722Config& config, int payload_type);
+  ~AudioEncoderG722Impl() override;
+
+  AudioEncoderG722Impl(const AudioEncoderG722Impl&) = delete;
+  AudioEncoderG722Impl& operator=(const AudioEncoderG722Impl&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  // The encoder state for one channel.
+  struct EncoderState {
+    G722EncInst* encoder;
+    std::unique_ptr<int16_t[]> speech_buffer;  // Queued up for encoding.
+    rtc::Buffer encoded_buffer;                // Already encoded.
+    EncoderState();
+    ~EncoderState();
+  };
+
+  size_t SamplesPerChannel() const;
+
+  const size_t num_channels_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  size_t num_10ms_frames_buffered_;
+  uint32_t first_timestamp_in_buffer_;
+  const std::unique_ptr<EncoderState[]> encoders_;
+  rtc::Buffer interleave_buffer_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.c b/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.c
new file mode 100644
index 0000000000..36ee6d92be
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.c
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "modules/third_party/g722/g722_enc_dec.h"
+
+int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
+{
+    *G722enc_inst=(G722EncInst*)malloc(sizeof(G722EncoderState));
+    if (*G722enc_inst!=NULL) {
+      return(0);
+    } else {
+      return(-1);
+    }
+}
+
+int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
+{
+    // Create and/or reset the G.722 encoder
+    // Bitrate 64 kbps and wideband mode (2)
+    G722enc_inst = (G722EncInst *) WebRtc_g722_encode_init(
+        (G722EncoderState*) G722enc_inst, 64000, 2);
+    if (G722enc_inst == NULL) {
+        return -1;
+    } else {
+        return 0;
+    }
+}
+
+int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
+{
+    // Free encoder memory
+    return WebRtc_g722_encode_release((G722EncoderState*) G722enc_inst);
+}
+
+size_t WebRtcG722_Encode(G722EncInst *G722enc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded)
+{
+    unsigned char *codechar = (unsigned char*) encoded;
+    // Encode the input speech vector
+    return WebRtc_g722_encode((G722EncoderState*) G722enc_inst, codechar,
+                              speechIn, len);
+}
+
+int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
+{
+    *G722dec_inst=(G722DecInst*)malloc(sizeof(G722DecoderState));
+    if (*G722dec_inst!=NULL) {
+      return(0);
+    } else {
+      return(-1);
+    }
+}
+
+void WebRtcG722_DecoderInit(G722DecInst* inst) {
+  // Create and/or reset the G.722 decoder
+  // Bitrate 64 kbps and wideband mode (2)
+  WebRtc_g722_decode_init((G722DecoderState*)inst, 64000, 2);
+}
+
+int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
+{
+    // Free encoder memory
+    return WebRtc_g722_decode_release((G722DecoderState*) G722dec_inst);
+}
+
+size_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
+                         const uint8_t *encoded,
+                         size_t len,
+                         int16_t *decoded,
+                         int16_t *speechType)
+{
+    // Decode the G.722 encoder stream
+    *speechType=G722_WEBRTC_SPEECH;
+    return WebRtc_g722_decode((G722DecoderState*) G722dec_inst, decoded,
+                              encoded, len);
+}
+
+int16_t WebRtcG722_Version(char *versionStr, short len)
+{
+    // Get version string
+    char version[30] = "2.0.0\n";
+    if (strlen(version) < (unsigned int)len)
+    {
+        strcpy(versionStr, version);
+        return 0;
+    }
+    else
+    {
+        return -1;
+    }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.h
new file mode 100644
index 0000000000..353de4504f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/g722_interface.h
@@ -0,0 +1,174 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*
+ * Solution to support multiple instances
+ */
+
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+/*
+ * Comfort noise constants
+ */
+
+#define G722_WEBRTC_SPEECH 1
+#define G722_WEBRTC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcG722_CreateEncoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance for encoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+int16_t WebRtcG722_CreateEncoder(G722EncInst** G722enc_inst);
+
+/****************************************************************************
+ * WebRtcG722_EncoderInit(...)
+ *
+ * This function initializes a G722 instance
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance, i.e. the user that should receive
+ *                             be initialized
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+
+int16_t WebRtcG722_EncoderInit(G722EncInst* G722enc_inst);
+
+/****************************************************************************
+ * WebRtcG722_FreeEncoder(...)
+ *
+ * Free the memory used for G722 encoder
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance for encoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+int WebRtcG722_FreeEncoder(G722EncInst* G722enc_inst);
+
+/****************************************************************************
+ * WebRtcG722_Encode(...)
+ *
+ * This function encodes G722 encoded data.
+ *
+ * Input:
+ *     - G722enc_inst         : G722 instance, i.e. the user that should encode
+ *                              a packet
+ *     - speechIn             : Input speech vector
+ *     - len                  : Samples in speechIn
+ *
+ * Output:
+ *        - encoded           : The encoded data vector
+ *
+ * Return value               : Length (in bytes) of coded data
+ */
+
+size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcG722_CreateDecoder(...)
+ *
+ * Create memory used for G722 encoder
+ *
+ * Input:
+ *     - G722dec_inst         : G722 instance for decoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+int16_t WebRtcG722_CreateDecoder(G722DecInst** G722dec_inst);
+
+/****************************************************************************
+ * WebRtcG722_DecoderInit(...)
+ *
+ * This function initializes a G722 instance
+ *
+ * Input:
+ *     - inst      : G722 instance
+ */
+
+void WebRtcG722_DecoderInit(G722DecInst* inst);
+
+/****************************************************************************
+ * WebRtcG722_FreeDecoder(...)
+ *
+ * Free the memory used for G722 decoder
+ *
+ * Input:
+ *     - G722dec_inst         : G722 instance for decoder
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+
+int WebRtcG722_FreeDecoder(G722DecInst* G722dec_inst);
+
+/****************************************************************************
+ * WebRtcG722_Decode(...)
+ *
+ * This function decodes a packet with G729 frame(s). Output speech length
+ * will be a multiple of 80 samples (80*frames/packet).
+ *
+ * Input:
+ *     - G722dec_inst       : G722 instance, i.e. the user that should decode
+ *                            a packet
+ *     - encoded            : Encoded G722 frame(s)
+ *     - len                : Bytes in encoded vector
+ *
+ * Output:
+ *        - decoded         : The decoded vector
+ *      - speechType        : 1 normal, 2 CNG (Since G722 does not have its own
+ *                            DTX/CNG scheme it should always return 1)
+ *
+ * Return value             : Samples in decoded vector
+ */
+
+size_t WebRtcG722_Decode(G722DecInst* G722dec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType);
+
+/****************************************************************************
+ * WebRtcG722_Version(...)
+ *
+ * Get a string with the current version of the codec
+ */
+
+int16_t WebRtcG722_Version(char* versionStr, short len);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/test/testG722.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g722/test/testG722.cc
new file mode 100644
index 0000000000..9f2155d0f7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/test/testG722.cc
@@ -0,0 +1,155 @@
+/*
+ *  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.
+ */
+
+/*
+ * testG722.cpp : Defines the entry point for the console application.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+
+/* include API */
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+
+/* Runtime statistics */
+#include <time.h>
+#define CLOCKS_PER_SEC_G722 100000
+
+// Forward declaration
+typedef struct WebRtcG722EncInst G722EncInst;
+typedef struct WebRtcG722DecInst G722DecInst;
+
+/* function for reading audio data from PCM file */
+bool readframe(int16_t* data, FILE* inp, size_t length) {
+  size_t rlen = fread(data, sizeof(int16_t), length, inp);
+  if (rlen >= length)
+    return false;
+  memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
+  return true;
+}
+
+int main(int argc, char* argv[]) {
+  char inname[60], outbit[40], outname[40];
+  FILE *inp, *outbitp, *outp;
+
+  int framecnt;
+  bool endfile;
+  size_t framelength = 160;
+  G722EncInst* G722enc_inst;
+  G722DecInst* G722dec_inst;
+
+  /* Runtime statistics */
+  double starttime;
+  double runtime = 0;
+  double length_file;
+
+  size_t stream_len = 0;
+  int16_t shortdata[960];
+  int16_t decoded[960];
+  uint8_t streamdata[80 * 6];
+  int16_t speechType[1];
+
+  /* handling wrong input arguments in the command line */
+  if (argc != 5) {
+    printf("\n\nWrong number of arguments or flag values.\n\n");
+
+    printf("\n");
+    printf("Usage:\n\n");
+    printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
+    printf("with:\n");
+    printf("framelength  :    Framelength in samples.\n\n");
+    printf("infile       :    Normal speech input file\n\n");
+    printf("outbitfile   :    Bitstream output file\n\n");
+    printf("outspeechfile:    Speech output file\n\n");
+    exit(0);
+  }
+
+  /* Get frame length */
+  int framelength_int = atoi(argv[1]);
+  if (framelength_int < 0) {
+    printf("  G.722: Invalid framelength %d.\n", framelength_int);
+    exit(1);
+  }
+  framelength = static_cast<size_t>(framelength_int);
+
+  /* Get Input and Output files */
+  sscanf(argv[2], "%s", inname);
+  sscanf(argv[3], "%s", outbit);
+  sscanf(argv[4], "%s", outname);
+
+  if ((inp = fopen(inname, "rb")) == NULL) {
+    printf("  G.722: Cannot read file %s.\n", inname);
+    exit(1);
+  }
+  if ((outbitp = fopen(outbit, "wb")) == NULL) {
+    printf("  G.722: Cannot write file %s.\n", outbit);
+    exit(1);
+  }
+  if ((outp = fopen(outname, "wb")) == NULL) {
+    printf("  G.722: Cannot write file %s.\n", outname);
+    exit(1);
+  }
+  printf("\nInput:%s\nOutput bitstream:%s\nOutput:%s\n", inname, outbit,
+         outname);
+
+  /* Create and init */
+  WebRtcG722_CreateEncoder((G722EncInst**)&G722enc_inst);
+  WebRtcG722_CreateDecoder((G722DecInst**)&G722dec_inst);
+  WebRtcG722_EncoderInit((G722EncInst*)G722enc_inst);
+  WebRtcG722_DecoderInit((G722DecInst*)G722dec_inst);
+
+  /* Initialize encoder and decoder */
+  framecnt = 0;
+  endfile = false;
+  while (!endfile) {
+    framecnt++;
+
+    /* Read speech block */
+    endfile = readframe(shortdata, inp, framelength);
+
+    /* Start clock before call to encoder and decoder */
+    starttime = clock() / (double)CLOCKS_PER_SEC_G722;
+
+    /* G.722 encoding + decoding */
+    stream_len = WebRtcG722_Encode((G722EncInst*)G722enc_inst, shortdata,
+                                   framelength, streamdata);
+    WebRtcG722_Decode(G722dec_inst, streamdata, stream_len, decoded,
+                      speechType);
+
+    /* Stop clock after call to encoder and decoder */
+    runtime += (double)((clock() / (double)CLOCKS_PER_SEC_G722) - starttime);
+
+    /* Write coded bits to file */
+    if (fwrite(streamdata, sizeof(short), stream_len / 2, outbitp) !=
+        stream_len / 2) {
+      return -1;
+    }
+    /* Write coded speech to file */
+    if (fwrite(decoded, sizeof(short), framelength, outp) != framelength) {
+      return -1;
+    }
+  }
+
+  WebRtcG722_FreeEncoder((G722EncInst*)G722enc_inst);
+  WebRtcG722_FreeDecoder((G722DecInst*)G722dec_inst);
+
+  length_file = ((double)framecnt * (double)framelength / 16000);
+  printf("\n\nLength of speech file: %.1f s\n", length_file);
+  printf("Time to run G.722:      %.2f s (%.2f %% of realtime)\n\n", runtime,
+         (100 * runtime / length_file));
+  printf("---------------------END----------------------\n");
+
+  fclose(inp);
+  fclose(outbitp);
+  fclose(outp);
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
new file mode 100644
index 0000000000..77da78ba7f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuant.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant.h"
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant_loop.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuant(
+    IlbcEncoder *iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
+                                   and idxVec, uses state_first as
+                                   input) */
+    int16_t *in,     /* (i) vector to encode */
+    int16_t *weightDenum   /* (i) denominator of synthesis filter */
+                            ) {
+  int16_t *syntOut;
+  size_t quantLen[2];
+
+  /* Stack based */
+  int16_t syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
+  int16_t in_weightedVec[STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t *in_weighted = &in_weightedVec[LPC_FILTERORDER];
+
+  /* Initialize the buffers */
+  WebRtcSpl_MemSetW16(syntOutBuf, 0, LPC_FILTERORDER+STATE_SHORT_LEN_30MS);
+  syntOut = &syntOutBuf[LPC_FILTERORDER];
+  /* Start with zero state */
+  WebRtcSpl_MemSetW16(in_weightedVec, 0, LPC_FILTERORDER);
+
+  /* Perform the quantization loop in two sections of length quantLen[i],
+     where the perceptual weighting filter is updated at the subframe
+     border */
+
+  if (iLBC_encbits->state_first) {
+    quantLen[0]=SUBL;
+    quantLen[1]=iLBCenc_inst->state_short_len-SUBL;
+  } else {
+    quantLen[0]=iLBCenc_inst->state_short_len-SUBL;
+    quantLen[1]=SUBL;
+  }
+
+  /* Calculate the weighted residual, switch perceptual weighting
+     filter at the subframe border */
+  WebRtcSpl_FilterARFastQ12(
+      in, in_weighted,
+      weightDenum, LPC_FILTERORDER+1, quantLen[0]);
+  WebRtcSpl_FilterARFastQ12(
+      &in[quantLen[0]], &in_weighted[quantLen[0]],
+      &weightDenum[LPC_FILTERORDER+1], LPC_FILTERORDER+1, quantLen[1]);
+
+  WebRtcIlbcfix_AbsQuantLoop(
+      syntOut,
+      in_weighted,
+      weightDenum,
+      quantLen,
+      iLBC_encbits->idxVec);
+
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
new file mode 100644
index 0000000000..c72e29cf29
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuant.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuant(
+    IlbcEncoder* iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits* iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
+                                   and idxVec, uses state_first as
+                                   input) */
+    int16_t* in,             /* (i) vector to encode */
+    int16_t* weightDenum     /* (i) denominator of synthesis filter */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
new file mode 100644
index 0000000000..cf9266299d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuantLoop.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant_loop.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/sort_sq.h"
+
+void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
+                                int16_t *weightDenumIN, size_t *quantLenIN,
+                                int16_t *idxVecIN ) {
+  size_t k1, k2;
+  int16_t index;
+  int32_t toQW32;
+  int32_t toQ32;
+  int16_t tmp16a;
+  int16_t xq;
+
+  int16_t *syntOut   = syntOutIN;
+  int16_t *in_weighted  = in_weightedIN;
+  int16_t *weightDenum  = weightDenumIN;
+  size_t *quantLen  = quantLenIN;
+  int16_t *idxVec   = idxVecIN;
+
+  for(k1=0;k1<2;k1++) {
+    for(k2=0;k2<quantLen[k1];k2++){
+
+      /* Filter to get the predicted value */
+      WebRtcSpl_FilterARFastQ12(
+          syntOut, syntOut,
+          weightDenum, LPC_FILTERORDER+1, 1);
+
+      /* the quantizer */
+      toQW32 = (int32_t)(*in_weighted) - (int32_t)(*syntOut);
+
+      toQ32 = (((int32_t)toQW32)<<2);
+
+      if (toQ32 > 32767) {
+        toQ32 = (int32_t) 32767;
+      } else if (toQ32 < -32768) {
+        toQ32 = (int32_t) -32768;
+      }
+
+      /* Quantize the state */
+      if (toQW32<(-7577)) {
+        /* To prevent negative overflow */
+        index=0;
+      } else if (toQW32>8151) {
+        /* To prevent positive overflow */
+        index=7;
+      } else {
+        /* Find the best quantization index
+           (state_sq3Tbl is in Q13 and toQ is in Q11)
+        */
+        WebRtcIlbcfix_SortSq(&xq, &index,
+                             (int16_t)toQ32,
+                             WebRtcIlbcfix_kStateSq3, 8);
+      }
+
+      /* Store selected index */
+      (*idxVec++) = index;
+
+      /* Compute decoded sample and update of the prediction filter */
+      tmp16a = ((WebRtcIlbcfix_kStateSq3[index] + 2 ) >> 2);
+
+      *syntOut     = (int16_t) (tmp16a + (int32_t)(*in_weighted) - toQW32);
+
+      syntOut++; in_weighted++;
+    }
+    /* Update perceptual weighting filter at subframe border */
+    weightDenum += 11;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
new file mode 100644
index 0000000000..841d73b9fb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AbsQuantLoop.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  predictive noise shaping encoding of scaled start state
+ *  (subrutine for WebRtcIlbcfix_StateSearch)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AbsQuantLoop(int16_t* syntOutIN,
+                                int16_t* in_weightedIN,
+                                int16_t* weightDenumIN,
+                                size_t* quantLenIN,
+                                int16_t* idxVecIN);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
new file mode 100644
index 0000000000..57b5abbe23
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
@@ -0,0 +1,110 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+AudioDecoderIlbcImpl::AudioDecoderIlbcImpl() {
+  WebRtcIlbcfix_DecoderCreate(&dec_state_);
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+}
+
+AudioDecoderIlbcImpl::~AudioDecoderIlbcImpl() {
+  WebRtcIlbcfix_DecoderFree(dec_state_);
+}
+
+bool AudioDecoderIlbcImpl::HasDecodePlc() const {
+  return true;
+}
+
+int AudioDecoderIlbcImpl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 8000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+size_t AudioDecoderIlbcImpl::DecodePlc(size_t num_frames, int16_t* decoded) {
+  return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
+}
+
+void AudioDecoderIlbcImpl::Reset() {
+  WebRtcIlbcfix_Decoderinit30Ms(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderIlbcImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+  size_t bytes_per_frame;
+  int timestamps_per_frame;
+  if (payload.size() >= 950) {
+    RTC_LOG(LS_WARNING)
+        << "AudioDecoderIlbcImpl::ParsePayload: Payload too large";
+    return results;
+  }
+  if (payload.size() % 38 == 0) {
+    // 20 ms frames.
+    bytes_per_frame = 38;
+    timestamps_per_frame = 160;
+  } else if (payload.size() % 50 == 0) {
+    // 30 ms frames.
+    bytes_per_frame = 50;
+    timestamps_per_frame = 240;
+  } else {
+    RTC_LOG(LS_WARNING)
+        << "AudioDecoderIlbcImpl::ParsePayload: Invalid payload";
+    return results;
+  }
+
+  RTC_DCHECK_EQ(0, payload.size() % bytes_per_frame);
+  if (payload.size() == bytes_per_frame) {
+    std::unique_ptr<EncodedAudioFrame> frame(
+        new LegacyEncodedAudioFrame(this, std::move(payload)));
+    results.emplace_back(timestamp, 0, std::move(frame));
+  } else {
+    size_t byte_offset;
+    uint32_t timestamp_offset;
+    for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
+         byte_offset += bytes_per_frame,
+        timestamp_offset += timestamps_per_frame) {
+      std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
+          this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame)));
+      results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
+    }
+  }
+
+  return results;
+}
+
+int AudioDecoderIlbcImpl::SampleRateHz() const {
+  return 8000;
+}
+
+size_t AudioDecoderIlbcImpl::Channels() const {
+  return 1;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
new file mode 100644
index 0000000000..46ba755148
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+
+typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
+
+namespace webrtc {
+
+class AudioDecoderIlbcImpl final : public AudioDecoder {
+ public:
+  AudioDecoderIlbcImpl();
+  ~AudioDecoderIlbcImpl() override;
+
+  AudioDecoderIlbcImpl(const AudioDecoderIlbcImpl&) = delete;
+  AudioDecoderIlbcImpl& operator=(const AudioDecoderIlbcImpl&) = delete;
+
+  bool HasDecodePlc() const override;
+  size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  IlbcDecoderInstance* dec_state_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
new file mode 100644
index 0000000000..9fbf42ceeb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+
+#include <algorithm>
+#include <cstdint>
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+namespace {
+
+const int kSampleRateHz = 8000;
+
+int GetIlbcBitrate(int ptime) {
+  switch (ptime) {
+    case 20:
+    case 40:
+      // 38 bytes per frame of 20 ms => 15200 bits/s.
+      return 15200;
+    case 30:
+    case 60:
+      // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
+      return 13333;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+
+}  // namespace
+
+AudioEncoderIlbcImpl::AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config,
+                                           int payload_type)
+    : frame_size_ms_(config.frame_size_ms),
+      payload_type_(payload_type),
+      num_10ms_frames_per_packet_(
+          static_cast<size_t>(config.frame_size_ms / 10)),
+      encoder_(nullptr) {
+  RTC_CHECK(config.IsOk());
+  Reset();
+}
+
+AudioEncoderIlbcImpl::~AudioEncoderIlbcImpl() {
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+}
+
+int AudioEncoderIlbcImpl::SampleRateHz() const {
+  return kSampleRateHz;
+}
+
+size_t AudioEncoderIlbcImpl::NumChannels() const {
+  return 1;
+}
+
+size_t AudioEncoderIlbcImpl::Num10MsFramesInNextPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+size_t AudioEncoderIlbcImpl::Max10MsFramesInAPacket() const {
+  return num_10ms_frames_per_packet_;
+}
+
+int AudioEncoderIlbcImpl::GetTargetBitrate() const {
+  return GetIlbcBitrate(rtc::dchecked_cast<int>(num_10ms_frames_per_packet_) *
+                        10);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  // Save timestamp if starting a new packet.
+  if (num_10ms_frames_buffered_ == 0)
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  // Buffer input.
+  std::copy(audio.cbegin(), audio.cend(),
+            input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
+
+  // If we don't yet have enough buffered input for a whole packet, we're done
+  // for now.
+  if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
+    return EncodedInfo();
+  }
+
+  // Encode buffered input.
+  RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  num_10ms_frames_buffered_ = 0;
+  size_t encoded_bytes = encoded->AppendData(
+      RequiredOutputSizeBytes(), [&](rtc::ArrayView<uint8_t> encoded) {
+        const int r = WebRtcIlbcfix_Encode(
+            encoder_, input_buffer_,
+            kSampleRateHz / 100 * num_10ms_frames_per_packet_, encoded.data());
+        RTC_CHECK_GE(r, 0);
+
+        return static_cast<size_t>(r);
+      });
+
+  RTC_DCHECK_EQ(encoded_bytes, RequiredOutputSizeBytes());
+
+  EncodedInfo info;
+  info.encoded_bytes = encoded_bytes;
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.encoder_type = CodecType::kIlbc;
+  return info;
+}
+
+void AudioEncoderIlbcImpl::Reset() {
+  if (encoder_)
+    RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
+  const int encoder_frame_size_ms =
+      frame_size_ms_ > 30 ? frame_size_ms_ / 2 : frame_size_ms_;
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
+  num_10ms_frames_buffered_ = 0;
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderIlbcImpl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(num_10ms_frames_per_packet_ * 10),
+           TimeDelta::Millis(num_10ms_frames_per_packet_ * 10)}};
+}
+
+size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
+  switch (num_10ms_frames_per_packet_) {
+    case 2:
+      return 38;
+    case 3:
+      return 50;
+    case 4:
+      return 2 * 38;
+    case 6:
+      return 2 * 50;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
new file mode 100644
index 0000000000..c8dfa2ca6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+namespace webrtc {
+
+class AudioEncoderIlbcImpl final : public AudioEncoder {
+ public:
+  AudioEncoderIlbcImpl(const AudioEncoderIlbcConfig& config, int payload_type);
+  ~AudioEncoderIlbcImpl() override;
+
+  AudioEncoderIlbcImpl(const AudioEncoderIlbcImpl&) = delete;
+  AudioEncoderIlbcImpl& operator=(const AudioEncoderIlbcImpl&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ private:
+  size_t RequiredOutputSizeBytes() const;
+
+  static constexpr size_t kMaxSamplesPerPacket = 480;
+  const int frame_size_ms_;
+  const int payload_type_;
+  const size_t num_10ms_frames_per_packet_;
+  size_t num_10ms_frames_buffered_;
+  uint32_t first_timestamp_in_buffer_;
+  int16_t input_buffer_[kMaxSamplesPerPacket];
+  IlbcEncoderInstance* encoder_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
new file mode 100644
index 0000000000..c915a2f9f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AugmentedCbCorr.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/augmented_cb_corr.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_AugmentedCbCorr(
+    int16_t *target,   /* (i) Target vector */
+    int16_t *buffer,   /* (i) Memory buffer */
+    int16_t *interpSamples, /* (i) buffer with
+                                     interpolated samples */
+    int32_t *crossDot,  /* (o) The cross correlation between
+                                 the target and the Augmented
+                                 vector */
+    size_t low,    /* (i) Lag to start from (typically
+                             20) */
+    size_t high,   /* (i) Lag to end at (typically 39) */
+    int scale)   /* (i) Scale factor to use for
+                              the crossDot */
+{
+  size_t lagcount;
+  size_t ilow;
+  int16_t *targetPtr;
+  int32_t *crossDotPtr;
+  int16_t *iSPtr=interpSamples;
+
+  /* Calculate the correlation between the target and the
+     interpolated codebook. The correlation is calculated in
+     3 sections with the interpolated part in the middle */
+  crossDotPtr=crossDot;
+  for (lagcount=low; lagcount<=high; lagcount++) {
+
+    ilow = lagcount - 4;
+
+    /* Compute dot product for the first (lagcount-4) samples */
+    (*crossDotPtr) = WebRtcSpl_DotProductWithScale(target, buffer-lagcount, ilow, scale);
+
+    /* Compute dot product on the interpolated samples */
+    (*crossDotPtr) += WebRtcSpl_DotProductWithScale(target+ilow, iSPtr, 4, scale);
+    targetPtr = target + lagcount;
+    iSPtr += lagcount-ilow;
+
+    /* Compute dot product for the remaining samples */
+    (*crossDotPtr) += WebRtcSpl_DotProductWithScale(targetPtr, buffer-lagcount, SUBL-lagcount, scale);
+    crossDotPtr++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
new file mode 100644
index 0000000000..2e9612e51a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_AugmentedCbCorr.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Calculate correlation between target and Augmented codebooks
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_AugmentedCbCorr(
+    int16_t* target,        /* (i) Target vector */
+    int16_t* buffer,        /* (i) Memory buffer */
+    int16_t* interpSamples, /* (i) buffer with
+                                           interpolated samples */
+    int32_t* crossDot,      /* (o) The cross correlation between
+                                               the target and the Augmented
+                                               vector */
+    size_t low,             /* (i) Lag to start from (typically
+                                                            20) */
+    size_t high,            /* (i) Lag to end at (typically 39 */
+    int scale);             /* (i) Scale factor to use for the crossDot */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.c
new file mode 100644
index 0000000000..1a9b882adf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.c
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_BwExpand.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+/* The output is in the same domain as the input */
+void WebRtcIlbcfix_BwExpand(
+    int16_t *out, /* (o) the bandwidth expanded lpc coefficients */
+    int16_t *in,  /* (i) the lpc coefficients before bandwidth
+                                   expansion */
+    int16_t *coef, /* (i) the bandwidth expansion factor Q15 */
+    int16_t length /* (i) the length of lpc coefficient vectors */
+                            ) {
+  int i;
+
+  out[0] = in[0];
+  for (i = 1; i < length; i++) {
+    /* out[i] = coef[i] * in[i] with rounding.
+       in[] and out[] are in Q12 and coef[] is in Q15
+    */
+    out[i] = (int16_t)((coef[i] * in[i] + 16384) >> 15);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.h
new file mode 100644
index 0000000000..ff9b0b302e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/bw_expand.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_BwExpand.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  lpc bandwidth expansion
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_BwExpand(
+    int16_t* out,  /* (o) the bandwidth expanded lpc coefficients */
+    int16_t* in,   /* (i) the lpc coefficients before bandwidth
+                                    expansion */
+    int16_t* coef, /* (i) the bandwidth expansion factor Q15 */
+    int16_t length /* (i) the length of lpc coefficient vectors */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
new file mode 100644
index 0000000000..1e9a7040c7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbConstruct.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/gain_dequant.h"
+#include "modules/audio_coding/codecs/ilbc/get_cd_vec.h"
+#include "rtc_base/sanitizer.h"
+
+// An arithmetic operation that is allowed to overflow. (It's still undefined
+// behavior, so not a good idea; this just makes UBSan ignore the violation, so
+// that our old code can continue to do what it's always been doing.)
+static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
+    OverflowingAddS32S32ToS32(int32_t a, int32_t b) {
+  return a + b;
+}
+
+/*----------------------------------------------------------------*
+ *  Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+bool WebRtcIlbcfix_CbConstruct(
+    int16_t* decvector,        /* (o) Decoded vector */
+    const int16_t* index,      /* (i) Codebook indices */
+    const int16_t* gain_index, /* (i) Gain quantization indices */
+    int16_t* mem,              /* (i) Buffer for codevector construction */
+    size_t lMem,               /* (i) Length of buffer */
+    size_t veclen) {           /* (i) Length of vector */
+  size_t j;
+  int16_t gain[CB_NSTAGES];
+  /* Stack based */
+  int16_t cbvec0[SUBL];
+  int16_t cbvec1[SUBL];
+  int16_t cbvec2[SUBL];
+  int32_t a32;
+  int16_t *gainPtr;
+
+  /* gain de-quantization */
+
+  gain[0] = WebRtcIlbcfix_GainDequant(gain_index[0], 16384, 0);
+  gain[1] = WebRtcIlbcfix_GainDequant(gain_index[1], gain[0], 1);
+  gain[2] = WebRtcIlbcfix_GainDequant(gain_index[2], gain[1], 2);
+
+  /* codebook vector construction and construction of total vector */
+
+  /* Stack based */
+  if (!WebRtcIlbcfix_GetCbVec(cbvec0, mem, (size_t)index[0], lMem, veclen))
+    return false;  // Failure.
+  if (!WebRtcIlbcfix_GetCbVec(cbvec1, mem, (size_t)index[1], lMem, veclen))
+    return false;  // Failure.
+  if (!WebRtcIlbcfix_GetCbVec(cbvec2, mem, (size_t)index[2], lMem, veclen))
+    return false;  // Failure.
+
+  gainPtr = &gain[0];
+  for (j=0;j<veclen;j++) {
+    a32 = (*gainPtr++) * cbvec0[j];
+    a32 += (*gainPtr++) * cbvec1[j];
+    a32 = OverflowingAddS32S32ToS32(a32, (*gainPtr) * cbvec2[j]);
+    gainPtr -= 2;
+    decvector[j] = (int16_t)((a32 + 8192) >> 14);
+  }
+
+  return true;  // Success.
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
new file mode 100644
index 0000000000..8f7c663164
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbConstruct.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+
+#include <stdbool.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Construct decoded vector from codebook and gains.
+ *---------------------------------------------------------------*/
+
+// Returns true on success, false on failure.
+ABSL_MUST_USE_RESULT
+bool WebRtcIlbcfix_CbConstruct(
+    int16_t* decvector,        /* (o) Decoded vector */
+    const int16_t* index,      /* (i) Codebook indices */
+    const int16_t* gain_index, /* (i) Gain quantization indices */
+    int16_t* mem,              /* (i) Buffer for codevector construction */
+    size_t lMem,               /* (i) Length of buffer */
+    size_t veclen              /* (i) Length of vector */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
new file mode 100644
index 0000000000..21e4197607
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergy.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy.h"
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Function WebRtcIlbcfix_CbMemEnergy computes the energy of all
+ * the vectors in the codebook memory that will be used in the
+ * following search for the best match.
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbMemEnergy(
+    size_t range,
+    int16_t *CB,   /* (i) The CB memory (1:st section) */
+    int16_t *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
+    size_t lMem,   /* (i) Length of the CB memory */
+    size_t lTarget,   /* (i) Length of the target vector */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
+                               ) {
+  int16_t *ppi, *ppo, *pp;
+  int32_t energy, tmp32;
+
+  /* Compute the energy and store it in a vector. Also the
+   * corresponding shift values are stored. The energy values
+   * are reused in all three stages. */
+
+  /* Calculate the energy in the first block of 'lTarget' sampels. */
+  ppi = CB+lMem-lTarget-1;
+  ppo = CB+lMem-1;
+
+  pp=CB+lMem-lTarget;
+  energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
+
+  /* Normalize the energy and store the number of shifts */
+  energyShifts[0] = (int16_t)WebRtcSpl_NormW32(energy);
+  tmp32 = energy << energyShifts[0];
+  energyW16[0] = (int16_t)(tmp32 >> 16);
+
+  /* Compute the energy of the rest of the cb memory
+   * by step wise adding and subtracting the next
+   * sample and the last sample respectively. */
+  WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, 0);
+
+  /* Next, precompute the energy values for the filtered cb section */
+  energy=0;
+  pp=filteredCB+lMem-lTarget;
+
+  energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
+
+  /* Normalize the energy and store the number of shifts */
+  energyShifts[base_size] = (int16_t)WebRtcSpl_NormW32(energy);
+  tmp32 = energy << energyShifts[base_size];
+  energyW16[base_size] = (int16_t)(tmp32 >> 16);
+
+  ppi = filteredCB + lMem - 1 - lTarget;
+  ppo = filteredCB + lMem - 1;
+
+  WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, base_size);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
new file mode 100644
index 0000000000..17ec337dc6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergy.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbMemEnergy(
+    size_t range,
+    int16_t* CB,           /* (i) The CB memory (1:st section) */
+    int16_t* filteredCB,   /* (i) The filtered CB memory (2:nd section) */
+    size_t lMem,           /* (i) Length of the CB memory */
+    size_t lTarget,        /* (i) Length of the target vector */
+    int16_t* energyW16,    /* (o) Energy in the CB vectors */
+    int16_t* energyShifts, /* (o) Shift value of the energy */
+    int scale,             /* (i) The scaling of all energy values */
+    size_t base_size /* (i) Index to where energy values should be stored */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
new file mode 100644
index 0000000000..0619bbe422
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyAugmentation.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbMemEnergyAugmentation(
+    int16_t *interpSamples, /* (i) The interpolated samples */
+    int16_t *CBmem,   /* (i) The CB memory */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size,  /* (i) Index to where energy values should be stored */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts /* (o) Shift value of the energy */
+                                           ){
+  int32_t energy, tmp32;
+  int16_t *ppe, *pp, *interpSamplesPtr;
+  int16_t *CBmemPtr;
+  size_t lagcount;
+  int16_t *enPtr=&energyW16[base_size-20];
+  int16_t *enShPtr=&energyShifts[base_size-20];
+  int32_t nrjRecursive;
+
+  CBmemPtr = CBmem+147;
+  interpSamplesPtr = interpSamples;
+
+  /* Compute the energy for the first (low-5) noninterpolated samples */
+  nrjRecursive = WebRtcSpl_DotProductWithScale( CBmemPtr-19, CBmemPtr-19, 15, scale);
+  ppe = CBmemPtr - 20;
+
+  for (lagcount=20; lagcount<=39; lagcount++) {
+
+    /* Update the energy recursively to save complexity */
+    nrjRecursive += (*ppe * *ppe) >> scale;
+    ppe--;
+    energy = nrjRecursive;
+
+    /* interpolation */
+    energy += WebRtcSpl_DotProductWithScale(interpSamplesPtr, interpSamplesPtr, 4, scale);
+    interpSamplesPtr += 4;
+
+    /* Compute energy for the remaining samples */
+    pp = CBmemPtr - lagcount;
+    energy += WebRtcSpl_DotProductWithScale(pp, pp, SUBL-lagcount, scale);
+
+    /* Normalize the energy and store the number of shifts */
+    (*enShPtr) = (int16_t)WebRtcSpl_NormW32(energy);
+    tmp32 = energy << *enShPtr;
+    *enPtr = (int16_t)(tmp32 >> 16);
+    enShPtr++;
+    enPtr++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
new file mode 100644
index 0000000000..d7b7a0d97e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyAugmentation.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbMemEnergyAugmentation(
+    int16_t* interpSamples, /* (i) The interpolated samples */
+    int16_t* CBmem,         /* (i) The CB memory */
+    int scale,              /* (i) The scaling of all energy values */
+    size_t base_size,   /* (i) Index to where energy values should be stored */
+    int16_t* energyW16, /* (o) Energy in the CB vectors */
+    int16_t* energyShifts /* (o) Shift value of the energy */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
new file mode 100644
index 0000000000..58c0c5fe6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyCalc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* Compute the energy of the rest of the cb memory
+ * by step wise adding and subtracting the next
+ * sample and the last sample respectively */
+void WebRtcIlbcfix_CbMemEnergyCalc(
+    int32_t energy,   /* (i) input start energy */
+    size_t range,   /* (i) number of iterations */
+    int16_t *ppi,   /* (i) input pointer 1 */
+    int16_t *ppo,   /* (i) input pointer 2 */
+    int16_t *energyW16,  /* (o) Energy in the CB vectors */
+    int16_t *energyShifts, /* (o) Shift value of the energy */
+    int scale,   /* (i) The scaling of all energy values */
+    size_t base_size  /* (i) Index to where energy values should be stored */
+                                   )
+{
+  size_t j;
+  int16_t shft;
+  int32_t tmp;
+  int16_t *eSh_ptr;
+  int16_t *eW16_ptr;
+
+
+  eSh_ptr  = &energyShifts[1+base_size];
+  eW16_ptr = &energyW16[1+base_size];
+
+  for (j = 0; j + 1 < range; j++) {
+
+    /* Calculate next energy by a +/-
+       operation on the edge samples */
+    tmp = (*ppi) * (*ppi) - (*ppo) * (*ppo);
+    energy += tmp >> scale;
+    energy = WEBRTC_SPL_MAX(energy, 0);
+
+    ppi--;
+    ppo--;
+
+    /* Normalize the energy into a int16_t and store
+       the number of shifts */
+
+    shft = (int16_t)WebRtcSpl_NormW32(energy);
+    *eSh_ptr++ = shft;
+
+    tmp = energy << shft;
+    *eW16_ptr++ = (int16_t)(tmp >> 16);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
new file mode 100644
index 0000000000..1d1e8d62b9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbMemEnergyCalc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbMemEnergyCalc(
+    int32_t energy,        /* (i) input start energy */
+    size_t range,          /* (i) number of iterations */
+    int16_t* ppi,          /* (i) input pointer 1 */
+    int16_t* ppo,          /* (i) input pointer 2 */
+    int16_t* energyW16,    /* (o) Energy in the CB vectors */
+    int16_t* energyShifts, /* (o) Shift value of the energy */
+    int scale,             /* (i) The scaling of all energy values */
+    size_t base_size /* (i) Index to where energy values should be stored */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.c
new file mode 100644
index 0000000000..24b5292354
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -0,0 +1,405 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearch.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_search.h"
+
+#include "modules/audio_coding/codecs/ilbc/augmented_cb_corr.h"
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy.h"
+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h"
+#include "modules/audio_coding/codecs/ilbc/cb_search_core.h"
+#include "modules/audio_coding/codecs/ilbc/cb_update_best_index.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/energy_inverse.h"
+#include "modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h"
+#include "modules/audio_coding/codecs/ilbc/gain_quant.h"
+#include "modules/audio_coding/codecs/ilbc/interpolate_samples.h"
+
+/*----------------------------------------------------------------*
+ *  Search routine for codebook encoding and gain quantization.
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CbSearch(
+    IlbcEncoder *iLBCenc_inst,
+    /* (i) the encoder state structure */
+    int16_t *index,  /* (o) Codebook indices */
+    int16_t *gain_index, /* (o) Gain quantization indices */
+    int16_t *intarget, /* (i) Target vector for encoding */
+    int16_t *decResidual,/* (i) Decoded residual for codebook construction */
+    size_t lMem,  /* (i) Length of buffer */
+    size_t lTarget,  /* (i) Length of vector */
+    int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */
+    size_t block  /* (i) the subblock number */
+                            ) {
+  size_t i, range;
+  int16_t ii, j, stage;
+  int16_t *pp;
+  int16_t tmp;
+  int scale;
+  int16_t bits, temp1, temp2;
+  size_t base_size;
+  int32_t codedEner, targetEner;
+  int16_t gains[CB_NSTAGES+1];
+  int16_t *cb_vecPtr;
+  size_t indexOffset, sInd, eInd;
+  int32_t CritMax=0;
+  int16_t shTotMax=WEBRTC_SPL_WORD16_MIN;
+  size_t bestIndex=0;
+  int16_t bestGain=0;
+  size_t indexNew;
+  int16_t CritNewSh;
+  int32_t CritNew;
+  int32_t *cDotPtr;
+  size_t noOfZeros;
+  int16_t *gainPtr;
+  int32_t t32, tmpW32;
+  int16_t *WebRtcIlbcfix_kGainSq5_ptr;
+  /* Stack based */
+  int16_t CBbuf[CB_MEML+LPC_FILTERORDER+CB_HALFFILTERLEN];
+  int32_t cDot[128];
+  int32_t Crit[128];
+  int16_t targetVec[SUBL+LPC_FILTERORDER];
+  int16_t cbvectors[CB_MEML + 1];  /* Adding one extra position for
+                                            Coverity warnings. */
+  int16_t codedVec[SUBL];
+  int16_t interpSamples[20*4];
+  int16_t interpSamplesFilt[20*4];
+  int16_t energyW16[CB_EXPAND*128];
+  int16_t energyShifts[CB_EXPAND*128];
+  int16_t *inverseEnergy=energyW16;   /* Reuse memory */
+  int16_t *inverseEnergyShifts=energyShifts; /* Reuse memory */
+  int16_t *buf = &CBbuf[LPC_FILTERORDER];
+  int16_t *target = &targetVec[LPC_FILTERORDER];
+  int16_t *aug_vec = (int16_t*)cDot;   /* length [SUBL], reuse memory */
+
+  /* Determine size of codebook sections */
+
+  base_size=lMem-lTarget+1;
+  if (lTarget==SUBL) {
+    base_size=lMem-19;
+  }
+
+  /* weighting of the CB memory */
+  noOfZeros=lMem-WebRtcIlbcfix_kFilterRange[block];
+  WebRtcSpl_MemSetW16(&buf[-LPC_FILTERORDER], 0, noOfZeros+LPC_FILTERORDER);
+  WebRtcSpl_FilterARFastQ12(
+      decResidual+noOfZeros, buf+noOfZeros,
+      weightDenum, LPC_FILTERORDER+1, WebRtcIlbcfix_kFilterRange[block]);
+
+  /* weighting of the target vector */
+  WEBRTC_SPL_MEMCPY_W16(&target[-LPC_FILTERORDER], buf+noOfZeros+WebRtcIlbcfix_kFilterRange[block]-LPC_FILTERORDER, LPC_FILTERORDER);
+  WebRtcSpl_FilterARFastQ12(
+      intarget, target,
+      weightDenum, LPC_FILTERORDER+1, lTarget);
+
+  /* Store target, towards the end codedVec is calculated as
+     the initial target minus the remaining target */
+  WEBRTC_SPL_MEMCPY_W16(codedVec, target, lTarget);
+
+  /* Find the highest absolute value to calculate proper
+     vector scale factor (so that it uses 12 bits) */
+  temp1 = WebRtcSpl_MaxAbsValueW16(buf, lMem);
+  temp2 = WebRtcSpl_MaxAbsValueW16(target, lTarget);
+
+  if ((temp1>0)&&(temp2>0)) {
+    temp1 = WEBRTC_SPL_MAX(temp1, temp2);
+    scale = WebRtcSpl_GetSizeInBits((uint32_t)(temp1 * temp1));
+  } else {
+    /* temp1 or temp2 is negative (maximum was -32768) */
+    scale = 30;
+  }
+
+  /* Scale to so that a mul-add 40 times does not overflow */
+  scale = scale - 25;
+  scale = WEBRTC_SPL_MAX(0, scale);
+
+  /* Compute energy of the original target */
+  targetEner = WebRtcSpl_DotProductWithScale(target, target, lTarget, scale);
+
+  /* Prepare search over one more codebook section. This section
+     is created by filtering the original buffer with a filter. */
+  WebRtcIlbcfix_FilteredCbVecs(cbvectors, buf, lMem, WebRtcIlbcfix_kFilterRange[block]);
+
+  range = WebRtcIlbcfix_kSearchRange[block][0];
+
+  if(lTarget == SUBL) {
+    /* Create the interpolated samples and store them for use in all stages */
+
+    /* First section, non-filtered half of the cb */
+    WebRtcIlbcfix_InterpolateSamples(interpSamples, buf, lMem);
+
+    /* Second section, filtered half of the cb */
+    WebRtcIlbcfix_InterpolateSamples(interpSamplesFilt, cbvectors, lMem);
+
+    /* Compute the CB vectors' energies for the first cb section (non-filtered) */
+    WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamples, buf,
+                                          scale, 20, energyW16, energyShifts);
+
+    /* Compute the CB vectors' energies for the second cb section (filtered cb) */
+    WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamplesFilt, cbvectors, scale,
+                                          base_size + 20, energyW16,
+                                          energyShifts);
+
+    /* Compute the CB vectors' energies and store them in the vector
+     * energyW16. Also the corresponding shift values are stored. The
+     * energy values are used in all three stages. */
+    WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem,
+                              lTarget, energyW16+20, energyShifts+20, scale, base_size);
+
+  } else {
+    /* Compute the CB vectors' energies and store them in the vector
+     * energyW16. Also the corresponding shift values are stored. The
+     * energy values are used in all three stages. */
+    WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem,
+                              lTarget, energyW16, energyShifts, scale, base_size);
+
+    /* Set the energy positions 58-63 and 122-127 to zero
+       (otherwise they are uninitialized) */
+    WebRtcSpl_MemSetW16(energyW16+range, 0, (base_size-range));
+    WebRtcSpl_MemSetW16(energyW16+range+base_size, 0, (base_size-range));
+  }
+
+  /* Calculate Inverse Energy (energyW16 is already normalized
+     and will contain the inverse energy in Q29 after this call */
+  WebRtcIlbcfix_EnergyInverse(energyW16, base_size*CB_EXPAND);
+
+  /* The gain value computed in the previous stage is used
+   * as an upper limit to what the next stage gain value
+   * is allowed to be. In stage 0, 16384 (1.0 in Q14) is used as
+   * the upper limit. */
+  gains[0] = 16384;
+
+  for (stage=0; stage<CB_NSTAGES; stage++) {
+
+    /* Set up memories */
+    range = WebRtcIlbcfix_kSearchRange[block][stage];
+
+    /* initialize search measures */
+    CritMax=0;
+    shTotMax=-100;
+    bestIndex=0;
+    bestGain=0;
+
+    /* loop over lags 40+ in the first codebook section, full search */
+    cb_vecPtr = buf+lMem-lTarget;
+
+    /* Calculate all the cross correlations (augmented part of CB) */
+    if (lTarget==SUBL) {
+      WebRtcIlbcfix_AugmentedCbCorr(target, buf+lMem,
+                                    interpSamples, cDot,
+                                    20, 39, scale);
+      cDotPtr=&cDot[20];
+    } else {
+      cDotPtr=cDot;
+    }
+    /* Calculate all the cross correlations (main part of CB) */
+    WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, range, scale, -1);
+
+    /* Adjust the search range for the augmented vectors */
+    if (lTarget==SUBL) {
+      range=WebRtcIlbcfix_kSearchRange[block][stage]+20;
+    } else {
+      range=WebRtcIlbcfix_kSearchRange[block][stage];
+    }
+
+    indexOffset=0;
+
+    /* Search for best index in this part of the vector */
+    WebRtcIlbcfix_CbSearchCore(
+        cDot, range, stage, inverseEnergy,
+        inverseEnergyShifts, Crit,
+        &indexNew, &CritNew, &CritNewSh);
+
+    /* Update the global best index and the corresponding gain */
+    WebRtcIlbcfix_CbUpdateBestIndex(
+        CritNew, CritNewSh, indexNew+indexOffset, cDot[indexNew+indexOffset],
+        inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
+        &CritMax, &shTotMax, &bestIndex, &bestGain);
+
+    sInd = ((CB_RESRANGE >> 1) > bestIndex) ?
+        0 : (bestIndex - (CB_RESRANGE >> 1));
+    eInd=sInd+CB_RESRANGE;
+    if (eInd>=range) {
+      eInd=range-1;
+      sInd=eInd-CB_RESRANGE;
+    }
+
+    range = WebRtcIlbcfix_kSearchRange[block][stage];
+
+    if (lTarget==SUBL) {
+      i=sInd;
+      if (sInd<20) {
+        WebRtcIlbcfix_AugmentedCbCorr(target, cbvectors + lMem,
+                                      interpSamplesFilt, cDot, sInd + 20,
+                                      WEBRTC_SPL_MIN(39, (eInd + 20)), scale);
+        i=20;
+        cDotPtr = &cDot[20 - sInd];
+      } else {
+        cDotPtr = cDot;
+      }
+
+      cb_vecPtr = cbvectors+lMem-20-i;
+
+      /* Calculate the cross correlations (main part of the filtered CB) */
+      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget,
+                                 eInd - i + 1, scale, -1);
+
+    } else {
+      cDotPtr = cDot;
+      cb_vecPtr = cbvectors+lMem-lTarget-sInd;
+
+      /* Calculate the cross correlations (main part of the filtered CB) */
+      WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget,
+                                 eInd - sInd + 1, scale, -1);
+
+    }
+
+    /* Adjust the search range for the augmented vectors */
+    indexOffset=base_size+sInd;
+
+    /* Search for best index in this part of the vector */
+    WebRtcIlbcfix_CbSearchCore(
+        cDot, eInd-sInd+1, stage, inverseEnergy+indexOffset,
+        inverseEnergyShifts+indexOffset, Crit,
+        &indexNew, &CritNew, &CritNewSh);
+
+    /* Update the global best index and the corresponding gain */
+    WebRtcIlbcfix_CbUpdateBestIndex(
+        CritNew, CritNewSh, indexNew+indexOffset, cDot[indexNew],
+        inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
+        &CritMax, &shTotMax, &bestIndex, &bestGain);
+
+    index[stage] = (int16_t)bestIndex;
+
+
+    bestGain = WebRtcIlbcfix_GainQuant(bestGain,
+                                       (int16_t)WEBRTC_SPL_ABS_W16(gains[stage]), stage, &gain_index[stage]);
+
+    /* Extract the best (according to measure) codebook vector
+       Also adjust the index, so that the augmented vectors are last.
+       Above these vectors were first...
+    */
+
+    if(lTarget==(STATE_LEN-iLBCenc_inst->state_short_len)) {
+
+      if((size_t)index[stage]<base_size) {
+        pp=buf+lMem-lTarget-index[stage];
+      } else {
+        pp=cbvectors+lMem-lTarget-
+            index[stage]+base_size;
+      }
+
+    } else {
+
+      if ((size_t)index[stage]<base_size) {
+        if (index[stage]>=20) {
+          /* Adjust index and extract vector */
+          index[stage]-=20;
+          pp=buf+lMem-lTarget-index[stage];
+        } else {
+          /* Adjust index and extract vector */
+          index[stage]+=(int16_t)(base_size-20);
+
+          WebRtcIlbcfix_CreateAugmentedVec(index[stage]-base_size+40,
+                                           buf+lMem, aug_vec);
+          pp = aug_vec;
+
+        }
+      } else {
+
+        if ((index[stage] - base_size) >= 20) {
+          /* Adjust index and extract vector */
+          index[stage]-=20;
+          pp=cbvectors+lMem-lTarget-
+              index[stage]+base_size;
+        } else {
+          /* Adjust index and extract vector */
+          index[stage]+=(int16_t)(base_size-20);
+          WebRtcIlbcfix_CreateAugmentedVec(index[stage]-2*base_size+40,
+                                           cbvectors+lMem, aug_vec);
+          pp = aug_vec;
+        }
+      }
+    }
+
+    /* Subtract the best codebook vector, according
+       to measure, from the target vector */
+
+    WebRtcSpl_AddAffineVectorToVector(target, pp, (int16_t)(-bestGain),
+                                      (int32_t)8192, (int16_t)14, lTarget);
+
+    /* record quantized gain */
+    gains[stage+1] = bestGain;
+
+  } /* end of Main Loop. for (stage=0;... */
+
+  /* Calculte the coded vector (original target - what's left) */
+  for (i=0;i<lTarget;i++) {
+    codedVec[i]-=target[i];
+  }
+
+  /* Gain adjustment for energy matching */
+  codedEner = WebRtcSpl_DotProductWithScale(codedVec, codedVec, lTarget, scale);
+
+  j=gain_index[0];
+
+  temp1 = (int16_t)WebRtcSpl_NormW32(codedEner);
+  temp2 = (int16_t)WebRtcSpl_NormW32(targetEner);
+
+  if(temp1 < temp2) {
+    bits = 16 - temp1;
+  } else {
+    bits = 16 - temp2;
+  }
+
+  tmp = (int16_t)((gains[1] * gains[1]) >> 14);
+
+  targetEner = (int16_t)WEBRTC_SPL_SHIFT_W32(targetEner, -bits) * tmp;
+
+  tmpW32 = ((int32_t)(gains[1]-1))<<1;
+
+  /* Pointer to the table that contains
+     gain_sq5TblFIX * gain_sq5TblFIX in Q14 */
+  gainPtr=(int16_t*)WebRtcIlbcfix_kGainSq5Sq+gain_index[0];
+  temp1 = (int16_t)WEBRTC_SPL_SHIFT_W32(codedEner, -bits);
+
+  WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[j];
+
+  /* targetEner and codedEner are in Q(-2*scale) */
+  for (ii=gain_index[0];ii<32;ii++) {
+
+    /* Change the index if
+       (codedEnergy*gainTbl[i]*gainTbl[i])<(targetEn*gain[0]*gain[0]) AND
+       gainTbl[i] < 2*gain[0]
+    */
+
+    t32 = temp1 * *gainPtr;
+    t32 = t32 - targetEner;
+    if (t32 < 0) {
+      if ((*WebRtcIlbcfix_kGainSq5_ptr) < tmpW32) {
+        j=ii;
+        WebRtcIlbcfix_kGainSq5_ptr = (int16_t*)&WebRtcIlbcfix_kGainSq5[ii];
+      }
+    }
+    gainPtr++;
+  }
+  gain_index[0]=j;
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.h
new file mode 100644
index 0000000000..84a52c7868
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearch.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbSearch(
+    IlbcEncoder* iLBCenc_inst,
+    /* (i) the encoder state structure */
+    int16_t* index,       /* (o) Codebook indices */
+    int16_t* gain_index,  /* (o) Gain quantization indices */
+    int16_t* intarget,    /* (i) Target vector for encoding */
+    int16_t* decResidual, /* (i) Decoded residual for codebook construction */
+    size_t lMem,          /* (i) Length of buffer */
+    size_t lTarget,       /* (i) Length of vector */
+    int16_t* weightDenum, /* (i) weighting filter coefficients in Q12 */
+    size_t block          /* (i) the subblock number */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
new file mode 100644
index 0000000000..a75e5b0ab8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
@@ -0,0 +1,115 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearchCore.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_search_core.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbSearchCore(
+    int32_t *cDot,    /* (i) Cross Correlation */
+    size_t range,    /* (i) Search range */
+    int16_t stage,    /* (i) Stage of this search */
+    int16_t *inverseEnergy,  /* (i) Inversed energy */
+    int16_t *inverseEnergyShift, /* (i) Shifts of inversed energy
+                                           with the offset 2*16-29 */
+    int32_t *Crit,    /* (o) The criteria */
+    size_t *bestIndex,   /* (o) Index that corresponds to
+                                                   maximum criteria (in this
+                                                   vector) */
+    int32_t *bestCrit,   /* (o) Value of critera for the
+                                                   chosen index */
+    int16_t *bestCritSh)   /* (o) The domain of the chosen
+                                                   criteria */
+{
+  int32_t maxW32, tmp32;
+  int16_t max, sh, tmp16;
+  size_t i;
+  int32_t *cDotPtr;
+  int16_t cDotSqW16;
+  int16_t *inverseEnergyPtr;
+  int32_t *critPtr;
+  int16_t *inverseEnergyShiftPtr;
+
+  /* Don't allow negative values for stage 0 */
+  if (stage==0) {
+    cDotPtr=cDot;
+    for (i=0;i<range;i++) {
+      *cDotPtr=WEBRTC_SPL_MAX(0, (*cDotPtr));
+      cDotPtr++;
+    }
+  }
+
+  /* Normalize cDot to int16_t, calculate the square of cDot and store the upper int16_t */
+  maxW32 = WebRtcSpl_MaxAbsValueW32(cDot, range);
+
+  sh = (int16_t)WebRtcSpl_NormW32(maxW32);
+  cDotPtr = cDot;
+  inverseEnergyPtr = inverseEnergy;
+  critPtr = Crit;
+  inverseEnergyShiftPtr=inverseEnergyShift;
+  max=WEBRTC_SPL_WORD16_MIN;
+
+  for (i=0;i<range;i++) {
+    /* Calculate cDot*cDot and put the result in a int16_t */
+    tmp32 = *cDotPtr << sh;
+    tmp16 = (int16_t)(tmp32 >> 16);
+    cDotSqW16 = (int16_t)(((int32_t)(tmp16)*(tmp16))>>16);
+
+    /* Calculate the criteria (cDot*cDot/energy) */
+    *critPtr = cDotSqW16 * *inverseEnergyPtr;
+
+    /* Extract the maximum shift value under the constraint
+       that the criteria is not zero */
+    if ((*critPtr)!=0) {
+      max = WEBRTC_SPL_MAX((*inverseEnergyShiftPtr), max);
+    }
+
+    inverseEnergyPtr++;
+    inverseEnergyShiftPtr++;
+    critPtr++;
+    cDotPtr++;
+  }
+
+  /* If no max shifts still at initialization value, set shift to zero */
+  if (max==WEBRTC_SPL_WORD16_MIN) {
+    max = 0;
+  }
+
+  /* Modify the criterias, so that all of them use the same Q domain */
+  critPtr=Crit;
+  inverseEnergyShiftPtr=inverseEnergyShift;
+  for (i=0;i<range;i++) {
+    /* Guarantee that the shift value is less than 16
+       in order to simplify for DSP's (and guard against >31) */
+    tmp16 = WEBRTC_SPL_MIN(16, max-(*inverseEnergyShiftPtr));
+
+    (*critPtr)=WEBRTC_SPL_SHIFT_W32((*critPtr),-tmp16);
+    critPtr++;
+    inverseEnergyShiftPtr++;
+  }
+
+  /* Find the index of the best value */
+  *bestIndex = WebRtcSpl_MaxIndexW32(Crit, range);
+  *bestCrit = Crit[*bestIndex];
+
+  /* Calculate total shifts of this criteria */
+  *bestCritSh = 32 - 2*sh + max;
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
new file mode 100644
index 0000000000..5da70e0988
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbSearchCore.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbSearchCore(
+    int32_t* cDot,               /* (i) Cross Correlation */
+    size_t range,                /* (i) Search range */
+    int16_t stage,               /* (i) Stage of this search */
+    int16_t* inverseEnergy,      /* (i) Inversed energy */
+    int16_t* inverseEnergyShift, /* (i) Shifts of inversed energy
+                                          with the offset 2*16-29 */
+    int32_t* Crit,               /* (o) The criteria */
+    size_t* bestIndex,           /* (o) Index that corresponds to
+                                           maximum criteria (in this
+                                           vector) */
+    int32_t* bestCrit, /* (o) Value of critera for the
+                                chosen index */
+    int16_t* bestCritSh); /* (o) The domain of the chosen
+                                   criteria */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
new file mode 100644
index 0000000000..d6fa4d93d4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbUpdateBestIndex.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/cb_update_best_index.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_CbUpdateBestIndex(
+    int32_t CritNew,    /* (i) New Potentially best Criteria */
+    int16_t CritNewSh,   /* (i) Shift value of above Criteria */
+    size_t IndexNew,   /* (i) Index of new Criteria */
+    int32_t cDotNew,    /* (i) Cross dot of new index */
+    int16_t invEnergyNew,  /* (i) Inversed energy new index */
+    int16_t energyShiftNew,  /* (i) Energy shifts of new index */
+    int32_t *CritMax,   /* (i/o) Maximum Criteria (so far) */
+    int16_t *shTotMax,   /* (i/o) Shifts of maximum criteria */
+    size_t *bestIndex,   /* (i/o) Index that corresponds to
+                                                   maximum criteria */
+    int16_t *bestGain)   /* (i/o) Gain in Q14 that corresponds
+                                                   to maximum criteria */
+{
+  int16_t shOld, shNew, tmp16;
+  int16_t scaleTmp;
+  int32_t gainW32;
+
+  /* Normalize the new and old Criteria to the same domain */
+  if (CritNewSh>(*shTotMax)) {
+    shOld=WEBRTC_SPL_MIN(31,CritNewSh-(*shTotMax));
+    shNew=0;
+  } else {
+    shOld=0;
+    shNew=WEBRTC_SPL_MIN(31,(*shTotMax)-CritNewSh);
+  }
+
+  /* Compare the two criterias. If the new one is better,
+     calculate the gain and store this index as the new best one
+  */
+
+  if ((CritNew >> shNew) > (*CritMax >> shOld)) {
+
+    tmp16 = (int16_t)WebRtcSpl_NormW32(cDotNew);
+    tmp16 = 16 - tmp16;
+
+    /* Calculate the gain in Q14
+       Compensate for inverseEnergyshift in Q29 and that the energy
+       value was stored in a int16_t (shifted down 16 steps)
+       => 29-14+16 = 31 */
+
+    scaleTmp = -energyShiftNew-tmp16+31;
+    scaleTmp = WEBRTC_SPL_MIN(31, scaleTmp);
+
+    gainW32 = ((int16_t)WEBRTC_SPL_SHIFT_W32(cDotNew, -tmp16) * invEnergyNew) >>
+        scaleTmp;
+
+    /* Check if criteria satisfies Gain criteria (max 1.3)
+       if it is larger set the gain to 1.3
+       (slightly different from FLP version)
+    */
+    if (gainW32>21299) {
+      *bestGain=21299;
+    } else if (gainW32<-21299) {
+      *bestGain=-21299;
+    } else {
+      *bestGain=(int16_t)gainW32;
+    }
+
+    *CritMax=CritNew;
+    *shTotMax=CritNewSh;
+    *bestIndex = IndexNew;
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
new file mode 100644
index 0000000000..1a95d531e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CbUpdateBestIndex.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+void WebRtcIlbcfix_CbUpdateBestIndex(
+    int32_t CritNew,        /* (i) New Potentially best Criteria */
+    int16_t CritNewSh,      /* (i) Shift value of above Criteria */
+    size_t IndexNew,        /* (i) Index of new Criteria */
+    int32_t cDotNew,        /* (i) Cross dot of new index */
+    int16_t invEnergyNew,   /* (i) Inversed energy new index */
+    int16_t energyShiftNew, /* (i) Energy shifts of new index */
+    int32_t* CritMax,       /* (i/o) Maximum Criteria (so far) */
+    int16_t* shTotMax,      /* (i/o) Shifts of maximum criteria */
+    size_t* bestIndex,      /* (i/o) Index that corresponds to
+                                      maximum criteria */
+    int16_t* bestGain);     /* (i/o) Gain in Q14 that corresponds
+                                     to maximum criteria */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.c
new file mode 100644
index 0000000000..b4eee66219
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.c
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Chebyshev.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/chebyshev.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*------------------------------------------------------------------*
+ *  Calculate the Chevyshev polynomial series
+ *  F(w) = 2*exp(-j5w)*C(x)
+ *   C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
+ *   T_i(x) is the i:th order Chebyshev polynomial
+ *------------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_Chebyshev(
+    /* (o) Result of C(x) */
+    int16_t x,  /* (i) Value to the Chevyshev polynomial */
+    int16_t *f  /* (i) The coefficients in the polynomial */
+                                      ) {
+  int16_t b1_high, b1_low; /* Use the high, low format to increase the accuracy */
+  int32_t b2;
+  int32_t tmp1W32;
+  int32_t tmp2W32;
+  int i;
+
+  b2 = (int32_t)0x1000000; /* b2 = 1.0 (Q23) */
+  /* Calculate b1 = 2*x + f[1] */
+  tmp1W32 = (x << 10) + (f[1] << 14);
+
+  for (i = 2; i < 5; i++) {
+    tmp2W32 = tmp1W32;
+
+    /* Split b1 (in tmp1W32) into a high and low part */
+    b1_high = (int16_t)(tmp1W32 >> 16);
+    b1_low = (int16_t)((tmp1W32 - ((int32_t)b1_high << 16)) >> 1);
+
+    /* Calculate 2*x*b1-b2+f[i] */
+    tmp1W32 = ((b1_high * x + ((b1_low * x) >> 15)) << 2) - b2 + (f[i] << 14);
+
+    /* Update b2 for next round */
+    b2 = tmp2W32;
+  }
+
+  /* Split b1 (in tmp1W32) into a high and low part */
+  b1_high = (int16_t)(tmp1W32 >> 16);
+  b1_low = (int16_t)((tmp1W32 - ((int32_t)b1_high << 16)) >> 1);
+
+  /* tmp1W32 = x*b1 - b2 + f[i]/2 */
+  tmp1W32 = ((b1_high * x) << 1) + (((b1_low * x) >> 15) << 1) -
+      b2 + (f[i] << 13);
+
+  /* Handle overflows and set to maximum or minimum int16_t instead */
+  if (tmp1W32>((int32_t)33553408)) {
+    return(WEBRTC_SPL_WORD16_MAX);
+  } else if (tmp1W32<((int32_t)-33554432)) {
+    return(WEBRTC_SPL_WORD16_MIN);
+  } else {
+    return (int16_t)(tmp1W32 >> 10);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.h
new file mode 100644
index 0000000000..7e7742c5cc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/chebyshev.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Chebyshev.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*------------------------------------------------------------------*
+ *  Calculate the Chevyshev polynomial series
+ *  F(w) = 2*exp(-j5w)*C(x)
+ *   C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
+ *   T_i(x) is the i:th order Chebyshev polynomial
+ *------------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_Chebyshev(
+    /* (o) Result of C(x) */
+    int16_t x, /* (i) Value to the Chevyshev polynomial */
+    int16_t* f /* (i) The coefficients in the polynomial */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
new file mode 100644
index 0000000000..452bc78e3b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CompCorr.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/comp_corr.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Compute cross correlation and pitch gain for pitch prediction
+ *  of last subframe at given lag.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CompCorr(
+    int32_t *corr, /* (o) cross correlation */
+    int32_t *ener, /* (o) energy */
+    int16_t *buffer, /* (i) signal buffer */
+    size_t lag,  /* (i) pitch lag */
+    size_t bLen, /* (i) length of buffer */
+    size_t sRange, /* (i) correlation search length */
+    int16_t scale /* (i) number of rightshifts to use */
+                            ){
+  int16_t *w16ptr;
+
+  w16ptr=&buffer[bLen-sRange-lag];
+
+  /* Calculate correlation and energy */
+  (*corr)=WebRtcSpl_DotProductWithScale(&buffer[bLen-sRange], w16ptr, sRange, scale);
+  (*ener)=WebRtcSpl_DotProductWithScale(w16ptr, w16ptr, sRange, scale);
+
+  /* For zero energy set the energy to 0 in order to avoid potential
+     problems for coming divisions */
+  if (*ener == 0) {
+    *corr = 0;
+    *ener = 1;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
new file mode 100644
index 0000000000..010c6a1ce5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CompCorr.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Compute cross correlation and pitch gain for pitch prediction
+ *  of last subframe at given lag.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CompCorr(int32_t* corr,   /* (o) cross correlation */
+                            int32_t* ener,   /* (o) energy */
+                            int16_t* buffer, /* (i) signal buffer */
+                            size_t lag,      /* (i) pitch lag */
+                            size_t bLen,     /* (i) length of buffer */
+                            size_t sRange,   /* (i) correlation search length */
+                            int16_t scale /* (i) number of rightshifts to use */
+                            );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m
new file mode 100644
index 0000000000..4bda83622f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/complexityMeasures.m
@@ -0,0 +1,57 @@
+% % Copyright(c) 2011 The WebRTC project authors.All Rights Reserved.%
+        % Use of this source code is governed by a BSD
+    -
+    style license % that can be found in the LICENSE file in the root of the source
+        % tree.An additional intellectual property rights grant can be found
+        % in the file PATENTS.All contributing project authors may
+        % be found in the AUTHORS file in the root of the source tree.%
+
+        clear;
+pack;
+%
+% Enter the path to YOUR executable and remember to define the perprocessor
+% variable PRINT_MIPS te get the instructions printed to the screen.
+%
+command = '!iLBCtest.exe 30 speechAndBGnoise.pcm out1.bit out1.pcm tlm10_30ms.dat';
+cout=' > st.txt';   %saves to matlab variable 'st'
+eval(strcat(command,cout));
+if(length(cout)>3)
+    load st.txt
+else
+    disp('No cout file to load')
+end
+
+% initialize vector to zero
+index = find(st(1:end,1)==-1);
+indexnonzero = find(st(1:end,1)>0);
+frames = length(index)-indexnonzero(1)+1;
+start = indexnonzero(1) - 1;
+functionOrder=max(st(:,2));
+new=zeros(frames,functionOrder);
+
+for i = 1:frames,
+    for j = index(start-1+i)+1:(index(start+i)-1),
+        new(i,st(j,2)) = new(i,st(j,2)) + st(j,1);
+    end
+end
+
+result=zeros(functionOrder,3);
+for i=1:functionOrder
+    nonzeroelements = find(new(1:end,i)>0);
+    result(i,1)=i;
+    
+    % Compute each function's mean complexity
+    % result(i,2)=(sum(new(nonzeroelements,i))/(length(nonzeroelements)*0.03))/1000000;
+    
+    % Compute each function's maximum complexity in encoding
+    % and decoding respectively and then add it together:
+    % result(i,3)=(max(new(1:end,i))/0.03)/1000000;
+    result(i,3)=(max(new(1:size(new,1)/2,i))/0.03)/1000000 + (max(new(size(new,1)/2+1:end,i))/0.03)/1000000;
+end
+
+result
+
+% Compute maximum complexity for a single frame (enc/dec separately and together)
+maxEncComplexityInAFrame = (max(sum(new(1:size(new,1)/2,:),2))/0.03)/1000000
+maxDecComplexityInAFrame = (max(sum(new(size(new,1)/2+1:end,:),2))/0.03)/1000000
+totalComplexity = maxEncComplexityInAFrame + maxDecComplexityInAFrame
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.c
new file mode 100644
index 0000000000..22f2acb330
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.c
@@ -0,0 +1,667 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* HP Filters {b[0] b[1] b[2] -a[1] -a[2]} */
+
+const int16_t WebRtcIlbcfix_kHpInCoefs[5] = {3798, -7596, 3798, 7807, -3733};
+const int16_t WebRtcIlbcfix_kHpOutCoefs[5] = {3849, -7699, 3849, 7918, -3833};
+
+/* Window in Q11 to window the energies of the 5 choises (3 for 20ms) in the choise for
+   the 80 sample start state
+*/
+const int16_t WebRtcIlbcfix_kStartSequenceEnrgWin[NSUB_MAX-1]= {
+  1638, 1843, 2048, 1843, 1638
+};
+
+/* LP Filter coeffs used for downsampling */
+const int16_t WebRtcIlbcfix_kLpFiltCoefs[FILTERORDER_DS_PLUS1]= {
+  -273, 512, 1297, 1696, 1297, 512, -273
+};
+
+/* Constants used in the LPC calculations */
+
+/* Hanning LPC window (in Q15) */
+const int16_t WebRtcIlbcfix_kLpcWin[BLOCKL_MAX] = {
+  6, 22, 50, 89, 139, 200, 272, 355, 449, 554, 669, 795,
+  932, 1079, 1237, 1405, 1583, 1771, 1969, 2177, 2395, 2622, 2858, 3104,
+  3359, 3622, 3894, 4175, 4464, 4761, 5066, 5379, 5699, 6026, 6361, 6702,
+  7050, 7404, 7764, 8130, 8502, 8879, 9262, 9649, 10040, 10436, 10836, 11240,
+  11647, 12058, 12471, 12887, 13306, 13726, 14148, 14572, 14997, 15423, 15850, 16277,
+  16704, 17131, 17558, 17983, 18408, 18831, 19252, 19672, 20089, 20504, 20916, 21325,
+  21730, 22132, 22530, 22924, 23314, 23698, 24078, 24452, 24821, 25185, 25542, 25893,
+  26238, 26575, 26906, 27230, 27547, 27855, 28156, 28450, 28734, 29011, 29279, 29538,
+  29788, 30029, 30261, 30483, 30696, 30899, 31092, 31275, 31448, 31611, 31764, 31906,
+  32037, 32158, 32268, 32367, 32456, 32533, 32600, 32655, 32700, 32733, 32755, 32767,
+  32767, 32755, 32733, 32700, 32655, 32600, 32533, 32456, 32367, 32268, 32158, 32037,
+  31906, 31764, 31611, 31448, 31275, 31092, 30899, 30696, 30483, 30261, 30029, 29788,
+  29538, 29279, 29011, 28734, 28450, 28156, 27855, 27547, 27230, 26906, 26575, 26238,
+  25893, 25542, 25185, 24821, 24452, 24078, 23698, 23314, 22924, 22530, 22132, 21730,
+  21325, 20916, 20504, 20089, 19672, 19252, 18831, 18408, 17983, 17558, 17131, 16704,
+  16277, 15850, 15423, 14997, 14572, 14148, 13726, 13306, 12887, 12471, 12058, 11647,
+  11240, 10836, 10436, 10040, 9649, 9262, 8879, 8502, 8130, 7764, 7404, 7050,
+  6702, 6361, 6026, 5699, 5379, 5066, 4761, 4464, 4175, 3894, 3622, 3359,
+  3104, 2858, 2622, 2395, 2177, 1969, 1771, 1583, 1405, 1237, 1079, 932,
+  795, 669, 554, 449, 355, 272, 200, 139, 89, 50, 22, 6
+};
+
+/* Asymmetric LPC window (in Q15)*/
+const int16_t WebRtcIlbcfix_kLpcAsymWin[BLOCKL_MAX] = {
+  2, 7, 15, 27, 42, 60, 81, 106, 135, 166, 201, 239,
+  280, 325, 373, 424, 478, 536, 597, 661, 728, 798, 872, 949,
+  1028, 1111, 1197, 1287, 1379, 1474, 1572, 1674, 1778, 1885, 1995, 2108,
+  2224, 2343, 2465, 2589, 2717, 2847, 2980, 3115, 3254, 3395, 3538, 3684,
+  3833, 3984, 4138, 4295, 4453, 4615, 4778, 4944, 5112, 5283, 5456, 5631,
+  5808, 5987, 6169, 6352, 6538, 6725, 6915, 7106, 7300, 7495, 7692, 7891,
+  8091, 8293, 8497, 8702, 8909, 9118, 9328, 9539, 9752, 9966, 10182, 10398,
+  10616, 10835, 11055, 11277, 11499, 11722, 11947, 12172, 12398, 12625, 12852, 13080,
+  13309, 13539, 13769, 14000, 14231, 14463, 14695, 14927, 15160, 15393, 15626, 15859,
+  16092, 16326, 16559, 16792, 17026, 17259, 17492, 17725, 17957, 18189, 18421, 18653,
+  18884, 19114, 19344, 19573, 19802, 20030, 20257, 20483, 20709, 20934, 21157, 21380,
+  21602, 21823, 22042, 22261, 22478, 22694, 22909, 23123, 23335, 23545, 23755, 23962,
+  24168, 24373, 24576, 24777, 24977, 25175, 25371, 25565, 25758, 25948, 26137, 26323,
+  26508, 26690, 26871, 27049, 27225, 27399, 27571, 27740, 27907, 28072, 28234, 28394,
+  28552, 28707, 28860, 29010, 29157, 29302, 29444, 29584, 29721, 29855, 29987, 30115,
+  30241, 30364, 30485, 30602, 30717, 30828, 30937, 31043, 31145, 31245, 31342, 31436,
+  31526, 31614, 31699, 31780, 31858, 31933, 32005, 32074, 32140, 32202, 32261, 32317,
+  32370, 32420, 32466, 32509, 32549, 32585, 32618, 32648, 32675, 32698, 32718, 32734,
+  32748, 32758, 32764, 32767, 32767, 32667, 32365, 31863, 31164, 30274, 29197, 27939,
+  26510, 24917, 23170, 21281, 19261, 17121, 14876, 12540, 10126, 7650, 5126, 2571
+};
+
+/* Lag window for LPC (Q31) */
+const int32_t WebRtcIlbcfix_kLpcLagWin[LPC_FILTERORDER + 1]={
+  2147483647,   2144885453,   2137754373,   2125918626,   2109459810,
+  2088483140,   2063130336,   2033564590,   1999977009,   1962580174,
+  1921610283};
+
+/* WebRtcIlbcfix_kLpcChirpSyntDenum vector in Q15 corresponding
+ * floating point vector {1 0.9025 0.9025^2 0.9025^3 ...}
+ */
+const int16_t WebRtcIlbcfix_kLpcChirpSyntDenum[LPC_FILTERORDER + 1] = {
+  32767, 29573, 26690, 24087,
+  21739, 19619, 17707, 15980,
+  14422, 13016, 11747};
+
+/* WebRtcIlbcfix_kLpcChirpWeightDenum in Q15 corresponding to
+ * floating point vector {1 0.4222 0.4222^2... }
+ */
+const int16_t WebRtcIlbcfix_kLpcChirpWeightDenum[LPC_FILTERORDER + 1] = {
+  32767, 13835, 5841, 2466, 1041, 440,
+  186, 78,  33,  14,  6};
+
+/* LSF quantization Q13 domain */
+const int16_t WebRtcIlbcfix_kLsfCb[64 * 3 + 128 * 3 + 128 * 4] = {
+  1273,       2238,       3696,
+  3199,       5309,       8209,
+  3606,       5671,       7829,
+  2815,       5262,       8778,
+  2608,       4027,       5493,
+  1582,       3076,       5945,
+  2983,       4181,       5396,
+  2437,       4322,       6902,
+  1861,       2998,       4613,
+  2007,       3250,       5214,
+  1388,       2459,       4262,
+  2563,       3805,       5269,
+  2036,       3522,       5129,
+  1935,       4025,       6694,
+  2744,       5121,       7338,
+  2810,       4248,       5723,
+  3054,       5405,       7745,
+  1449,       2593,       4763,
+  3411,       5128,       6596,
+  2484,       4659,       7496,
+  1668,       2879,       4818,
+  1812,       3072,       5036,
+  1638,       2649,       3900,
+  2464,       3550,       4644,
+  1853,       2900,       4158,
+  2458,       4163,       5830,
+  2556,       4036,       6254,
+  2703,       4432,       6519,
+  3062,       4953,       7609,
+  1725,       3703,       6187,
+  2221,       3877,       5427,
+  2339,       3579,       5197,
+  2021,       4633,       7037,
+  2216,       3328,       4535,
+  2961,       4739,       6667,
+  2807,       3955,       5099,
+  2788,       4501,       6088,
+  1642,       2755,       4431,
+  3341,       5282,       7333,
+  2414,       3726,       5727,
+  1582,       2822,       5269,
+  2259,       3447,       4905,
+  3117,       4986,       7054,
+  1825,       3491,       5542,
+  3338,       5736,       8627,
+  1789,       3090,       5488,
+  2566,       3720,       4923,
+  2846,       4682,       7161,
+  1950,       3321,       5976,
+  1834,       3383,       6734,
+  3238,       4769,       6094,
+  2031,       3978,       5903,
+  1877,       4068,       7436,
+  2131,       4644,       8296,
+  2764,       5010,       8013,
+  2194,       3667,       6302,
+  2053,       3127,       4342,
+  3523,       6595,      10010,
+  3134,       4457,       5748,
+  3142,       5819,       9414,
+  2223,       4334,       6353,
+  2022,       3224,       4822,
+  2186,       3458,       5544,
+  2552,       4757,       6870,
+  10905,      12917,      14578,
+  9503,      11485,      14485,
+  9518,      12494,      14052,
+  6222,       7487,       9174,
+  7759,       9186,      10506,
+  8315,      12755,      14786,
+  9609,      11486,      13866,
+  8909,      12077,      13643,
+  7369,       9054,      11520,
+  9408,      12163,      14715,
+  6436,       9911,      12843,
+  7109,       9556,      11884,
+  7557,      10075,      11640,
+  6482,       9202,      11547,
+  6463,       7914,      10980,
+  8611,      10427,      12752,
+  7101,       9676,      12606,
+  7428,      11252,      13172,
+  10197,      12955,      15842,
+  7487,      10955,      12613,
+  5575,       7858,      13621,
+  7268,      11719,      14752,
+  7476,      11744,      13795,
+  7049,       8686,      11922,
+  8234,      11314,      13983,
+  6560,      11173,      14984,
+  6405,       9211,      12337,
+  8222,      12054,      13801,
+  8039,      10728,      13255,
+  10066,      12733,      14389,
+  6016,       7338,      10040,
+  6896,       8648,      10234,
+  7538,       9170,      12175,
+  7327,      12608,      14983,
+  10516,      12643,      15223,
+  5538,       7644,      12213,
+  6728,      12221,      14253,
+  7563,       9377,      12948,
+  8661,      11023,      13401,
+  7280,       8806,      11085,
+  7723,       9793,      12333,
+  12225,      14648,      16709,
+  8768,      13389,      15245,
+  10267,      12197,      13812,
+  5301,       7078,      11484,
+  7100,      10280,      11906,
+  8716,      12555,      14183,
+  9567,      12464,      15434,
+  7832,      12305,      14300,
+  7608,      10556,      12121,
+  8913,      11311,      12868,
+  7414,       9722,      11239,
+  8666,      11641,      13250,
+  9079,      10752,      12300,
+  8024,      11608,      13306,
+  10453,      13607,      16449,
+  8135,       9573,      10909,
+  6375,       7741,      10125,
+  10025,      12217,      14874,
+  6985,      11063,      14109,
+  9296,      13051,      14642,
+  8613,      10975,      12542,
+  6583,      10414,      13534,
+  6191,       9368,      13430,
+  5742,       6859,       9260,
+  7723,       9813,      13679,
+  8137,      11291,      12833,
+  6562,       8973,      10641,
+  6062,       8462,      11335,
+  6928,       8784,      12647,
+  7501,       8784,      10031,
+  8372,      10045,      12135,
+  8191,       9864,      12746,
+  5917,       7487,      10979,
+  5516,       6848,      10318,
+  6819,       9899,      11421,
+  7882,      12912,      15670,
+  9558,      11230,      12753,
+  7752,       9327,      11472,
+  8479,       9980,      11358,
+  11418,      14072,      16386,
+  7968,      10330,      14423,
+  8423,      10555,      12162,
+  6337,      10306,      14391,
+  8850,      10879,      14276,
+  6750,      11885,      15710,
+  7037,       8328,       9764,
+  6914,       9266,      13476,
+  9746,      13949,      15519,
+  11032,      14444,      16925,
+  8032,      10271,      11810,
+  10962,      13451,      15833,
+  10021,      11667,      13324,
+  6273,       8226,      12936,
+  8543,      10397,      13496,
+  7936,      10302,      12745,
+  6769,       8138,      10446,
+  6081,       7786,      11719,
+  8637,      11795,      14975,
+  8790,      10336,      11812,
+  7040,       8490,      10771,
+  7338,      10381,      13153,
+  6598,       7888,       9358,
+  6518,       8237,      12030,
+  9055,      10763,      12983,
+  6490,      10009,      12007,
+  9589,      12023,      13632,
+  6867,       9447,      10995,
+  7930,       9816,      11397,
+  10241,      13300,      14939,
+  5830,       8670,      12387,
+  9870,      11915,      14247,
+  9318,      11647,      13272,
+  6721,      10836,      12929,
+  6543,       8233,       9944,
+  8034,      10854,      12394,
+  9112,      11787,      14218,
+  9302,      11114,      13400,
+  9022,      11366,      13816,
+  6962,      10461,      12480,
+  11288,      13333,      15222,
+  7249,       8974,      10547,
+  10566,      12336,      14390,
+  6697,      11339,      13521,
+  11851,      13944,      15826,
+  6847,       8381,      11349,
+  7509,       9331,      10939,
+  8029,       9618,      11909,
+  13973,      17644,      19647,      22474,
+  14722,      16522,      20035,      22134,
+  16305,      18179,      21106,      23048,
+  15150,      17948,      21394,      23225,
+  13582,      15191,      17687,      22333,
+  11778,      15546,      18458,      21753,
+  16619,      18410,      20827,      23559,
+  14229,      15746,      17907,      22474,
+  12465,      15327,      20700,      22831,
+  15085,      16799,      20182,      23410,
+  13026,      16935,      19890,      22892,
+  14310,      16854,      19007,      22944,
+  14210,      15897,      18891,      23154,
+  14633,      18059,      20132,      22899,
+  15246,      17781,      19780,      22640,
+  16396,      18904,      20912,      23035,
+  14618,      17401,      19510,      21672,
+  15473,      17497,      19813,      23439,
+  18851,      20736,      22323,      23864,
+  15055,      16804,      18530,      20916,
+  16490,      18196,      19990,      21939,
+  11711,      15223,      21154,      23312,
+  13294,      15546,      19393,      21472,
+  12956,      16060,      20610,      22417,
+  11628,      15843,      19617,      22501,
+  14106,      16872,      19839,      22689,
+  15655,      18192,      20161,      22452,
+  12953,      15244,      20619,      23549,
+  15322,      17193,      19926,      21762,
+  16873,      18676,      20444,      22359,
+  14874,      17871,      20083,      21959,
+  11534,      14486,      19194,      21857,
+  17766,      19617,      21338,      23178,
+  13404,      15284,      19080,      23136,
+  15392,      17527,      19470,      21953,
+  14462,      16153,      17985,      21192,
+  17734,      19750,      21903,      23783,
+  16973,      19096,      21675,      23815,
+  16597,      18936,      21257,      23461,
+  15966,      17865,      20602,      22920,
+  15416,      17456,      20301,      22972,
+  18335,      20093,      21732,      23497,
+  15548,      17217,      20679,      23594,
+  15208,      16995,      20816,      22870,
+  13890,      18015,      20531,      22468,
+  13211,      15377,      19951,      22388,
+  12852,      14635,      17978,      22680,
+  16002,      17732,      20373,      23544,
+  11373,      14134,      19534,      22707,
+  17329,      19151,      21241,      23462,
+  15612,      17296,      19362,      22850,
+  15422,      19104,      21285,      23164,
+  13792,      17111,      19349,      21370,
+  15352,      17876,      20776,      22667,
+  15253,      16961,      18921,      22123,
+  14108,      17264,      20294,      23246,
+  15785,      17897,      20010,      21822,
+  17399,      19147,      20915,      22753,
+  13010,      15659,      18127,      20840,
+  16826,      19422,      22218,      24084,
+  18108,      20641,      22695,      24237,
+  18018,      20273,      22268,      23920,
+  16057,      17821,      21365,      23665,
+  16005,      17901,      19892,      23016,
+  13232,      16683,      21107,      23221,
+  13280,      16615,      19915,      21829,
+  14950,      18575,      20599,      22511,
+  16337,      18261,      20277,      23216,
+  14306,      16477,      21203,      23158,
+  12803,      17498,      20248,      22014,
+  14327,      17068,      20160,      22006,
+  14402,      17461,      21599,      23688,
+  16968,      18834,      20896,      23055,
+  15070,      17157,      20451,      22315,
+  15419,      17107,      21601,      23946,
+  16039,      17639,      19533,      21424,
+  16326,      19261,      21745,      23673,
+  16489,      18534,      21658,      23782,
+  16594,      18471,      20549,      22807,
+  18973,      21212,      22890,      24278,
+  14264,      18674,      21123,      23071,
+  15117,      16841,      19239,      23118,
+  13762,      15782,      20478,      23230,
+  14111,      15949,      20058,      22354,
+  14990,      16738,      21139,      23492,
+  13735,      16971,      19026,      22158,
+  14676,      17314,      20232,      22807,
+  16196,      18146,      20459,      22339,
+  14747,      17258,      19315,      22437,
+  14973,      17778,      20692,      23367,
+  15715,      17472,      20385,      22349,
+  15702,      18228,      20829,      23410,
+  14428,      16188,      20541,      23630,
+  16824,      19394,      21365,      23246,
+  13069,      16392,      18900,      21121,
+  12047,      16640,      19463,      21689,
+  14757,      17433,      19659,      23125,
+  15185,      16930,      19900,      22540,
+  16026,      17725,      19618,      22399,
+  16086,      18643,      21179,      23472,
+  15462,      17248,      19102,      21196,
+  17368,      20016,      22396,      24096,
+  12340,      14475,      19665,      23362,
+  13636,      16229,      19462,      22728,
+  14096,      16211,      19591,      21635,
+  12152,      14867,      19943,      22301,
+  14492,      17503,      21002,      22728,
+  14834,      16788,      19447,      21411,
+  14650,      16433,      19326,      22308,
+  14624,      16328,      19659,      23204,
+  13888,      16572,      20665,      22488,
+  12977,      16102,      18841,      22246,
+  15523,      18431,      21757,      23738,
+  14095,      16349,      18837,      20947,
+  13266,      17809,      21088,      22839,
+  15427,      18190,      20270,      23143,
+  11859,      16753,      20935,      22486,
+  12310,      17667,      21736,      23319,
+  14021,      15926,      18702,      22002,
+  12286,      15299,      19178,      21126,
+  15703,      17491,      21039,      23151,
+  12272,      14018,      18213,      22570,
+  14817,      16364,      18485,      22598,
+  17109,      19683,      21851,      23677,
+  12657,      14903,      19039,      22061,
+  14713,      16487,      20527,      22814,
+  14635,      16726,      18763,      21715,
+  15878,      18550,      20718,      22906
+};
+
+const int16_t WebRtcIlbcfix_kLsfDimCb[LSF_NSPLIT] = {3, 3, 4};
+const int16_t WebRtcIlbcfix_kLsfSizeCb[LSF_NSPLIT] = {64,128,128};
+
+const int16_t WebRtcIlbcfix_kLsfMean[LPC_FILTERORDER] = {
+  2308,       3652,       5434,       7885,
+  10255,      12559,      15160,      17513,
+  20328,      22752};
+
+const int16_t WebRtcIlbcfix_kLspMean[LPC_FILTERORDER] = {
+  31476, 29565, 25819, 18725, 10276,
+  1236, -9049, -17600, -25884, -30618
+};
+
+/* Q14 */
+const int16_t WebRtcIlbcfix_kLsfWeight20ms[4] = {12288, 8192, 4096, 0};
+const int16_t WebRtcIlbcfix_kLsfWeight30ms[6] = {8192, 16384, 10923, 5461, 0, 0};
+
+/*
+   cos(x) in Q15
+   WebRtcIlbcfix_kCos[i] = cos(pi*i/64.0)
+   used in WebRtcIlbcfix_Lsp2Lsf()
+*/
+
+const int16_t WebRtcIlbcfix_kCos[64] = {
+  32767,  32729,  32610,  32413,  32138,  31786,  31357,  30853,
+  30274,  29622,  28899,  28106,  27246,  26320,  25330,  24279,
+  23170,  22006,  20788,  19520,  18205,  16846,  15447,  14010,
+  12540,  11039,   9512,   7962,   6393,   4808,   3212,   1608,
+  0,  -1608,  -3212,  -4808,  -6393,  -7962,  -9512, -11039,
+  -12540, -14010, -15447, -16846, -18205, -19520, -20788, -22006,
+  -23170, -24279, -25330, -26320, -27246, -28106, -28899, -29622,
+  -30274, -30853, -31357, -31786, -32138, -32413, -32610, -32729
+};
+
+/*
+   Derivative in Q19, used to interpolate between the
+   WebRtcIlbcfix_kCos[] values to get a more exact y = cos(x)
+*/
+const int16_t WebRtcIlbcfix_kCosDerivative[64] = {
+  -632,  -1893,  -3150,  -4399,  -5638,  -6863,  -8072,  -9261,
+  -10428, -11570, -12684, -13767, -14817, -15832, -16808, -17744,
+  -18637, -19486, -20287, -21039, -21741, -22390, -22986, -23526,
+  -24009, -24435, -24801, -25108, -25354, -25540, -25664, -25726,
+  -25726, -25664, -25540, -25354, -25108, -24801, -24435, -24009,
+  -23526, -22986, -22390, -21741, -21039, -20287, -19486, -18637,
+  -17744, -16808, -15832, -14817, -13767, -12684, -11570, -10428,
+  -9261,  -8072,  -6863,  -5638,  -4399,  -3150,  -1893,   -632};
+
+/*
+  Table in Q15, used for a2lsf conversion
+  WebRtcIlbcfix_kCosGrid[i] = cos((2*pi*i)/(float)(2*COS_GRID_POINTS));
+*/
+
+const int16_t WebRtcIlbcfix_kCosGrid[COS_GRID_POINTS + 1] = {
+  32760, 32723, 32588, 32364, 32051, 31651, 31164, 30591,
+  29935, 29196, 28377, 27481, 26509, 25465, 24351, 23170,
+  21926, 20621, 19260, 17846, 16384, 14876, 13327, 11743,
+  10125, 8480, 6812, 5126, 3425, 1714, 0, -1714, -3425,
+  -5126, -6812, -8480, -10125, -11743, -13327, -14876,
+  -16384, -17846, -19260, -20621, -21926, -23170, -24351,
+  -25465, -26509, -27481, -28377, -29196, -29935, -30591,
+  -31164, -31651, -32051, -32364, -32588, -32723, -32760
+};
+
+/*
+   Derivative of y = acos(x) in Q12
+   used in WebRtcIlbcfix_Lsp2Lsf()
+*/
+
+const int16_t WebRtcIlbcfix_kAcosDerivative[64] = {
+  -26887, -8812, -5323, -3813, -2979, -2444, -2081, -1811,
+  -1608, -1450, -1322, -1219, -1132, -1059, -998, -946,
+  -901, -861, -827, -797, -772, -750, -730, -713,
+  -699, -687, -677, -668, -662, -657, -654, -652,
+  -652, -654, -657, -662, -668, -677, -687, -699,
+  -713, -730, -750, -772, -797, -827, -861, -901,
+  -946, -998, -1059, -1132, -1219, -1322, -1450, -1608,
+  -1811, -2081, -2444, -2979, -3813, -5323, -8812, -26887
+};
+
+
+/* Tables for quantization of start state */
+
+/* State quantization tables */
+const int16_t WebRtcIlbcfix_kStateSq3[8] = { /* Values in Q13 */
+  -30473, -17838, -9257, -2537,
+  3639, 10893, 19958, 32636
+};
+
+/* This table defines the limits for the selection of the freqg
+   less or equal than value 0 => index = 0
+   less or equal than value k => index = k
+*/
+const int32_t WebRtcIlbcfix_kChooseFrgQuant[64] = {
+  118, 163, 222, 305, 425, 604,
+  851, 1174, 1617, 2222, 3080, 4191,
+  5525, 7215, 9193, 11540, 14397, 17604,
+  21204, 25209, 29863, 35720, 42531, 50375,
+  59162, 68845, 80108, 93754, 110326, 129488,
+  150654, 174328, 201962, 233195, 267843, 308239,
+  354503, 405988, 464251, 531550, 608652, 697516,
+  802526, 928793, 1080145, 1258120, 1481106, 1760881,
+  2111111, 2546619, 3078825, 3748642, 4563142, 5573115,
+  6887601, 8582108, 10797296, 14014513, 18625760, 25529599,
+  37302935, 58819185, 109782723, WEBRTC_SPL_WORD32_MAX
+};
+
+const int16_t WebRtcIlbcfix_kScale[64] = {
+  /* Values in Q16 */
+  29485, 25003, 21345, 18316, 15578, 13128, 10973, 9310, 7955,
+  6762, 5789, 4877, 4255, 3699, 3258, 2904, 2595, 2328,
+  2123, 1932, 1785, 1631, 1493, 1370, 1260, 1167, 1083,
+  /* Values in Q21 */
+  32081, 29611, 27262, 25229, 23432, 21803, 20226, 18883, 17609,
+  16408, 15311, 14327, 13390, 12513, 11693, 10919, 10163, 9435,
+  8739, 8100, 7424, 6813, 6192, 5648, 5122, 4639, 4207, 3798,
+  3404, 3048, 2706, 2348, 2036, 1713, 1393, 1087, 747
+};
+
+/*frgq in fixpoint, but already computed like this:
+  for(i=0; i<64; i++){
+  a = (pow(10,frgq[i])/4.5);
+  WebRtcIlbcfix_kFrgQuantMod[i] = round(a);
+  }
+
+  Value 0 :36 in Q8
+  37:58 in Q5
+  59:63 in Q3
+*/
+const int16_t WebRtcIlbcfix_kFrgQuantMod[64] = {
+  /* First 37 values in Q8 */
+  569, 671, 786, 916, 1077, 1278,
+  1529, 1802, 2109, 2481, 2898, 3440,
+  3943, 4535, 5149, 5778, 6464, 7208,
+  7904, 8682, 9397, 10285, 11240, 12246,
+  13313, 14382, 15492, 16735, 18131, 19693,
+  21280, 22912, 24624, 26544, 28432, 30488,
+  32720,
+  /* 22 values in Q5 */
+  4383, 4684, 5012, 5363, 5739, 6146,
+  6603, 7113, 7679, 8285, 9040, 9850,
+  10838, 11882, 13103, 14467, 15950, 17669,
+  19712, 22016, 24800, 28576,
+  /* 5 values in Q3 */
+  8240, 9792, 12040, 15440, 22472
+};
+
+/* Constants for codebook search and creation */
+
+/* Expansion filter to get additional cb section.
+ * Q12 and reversed compared to flp
+ */
+const int16_t WebRtcIlbcfix_kCbFiltersRev[CB_FILTERLEN]={
+  -140, 446, -755, 3302, 2922, -590, 343, -138};
+
+/* Weighting coefficients for short lags.
+ * [0.2 0.4 0.6 0.8] in Q15 */
+const int16_t WebRtcIlbcfix_kAlpha[4]={
+  6554, 13107, 19661, 26214};
+
+/* Ranges for search and filters at different subframes */
+
+const size_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]={
+  {58,58,58}, {108,44,44}, {108,108,108}, {108,108,108}, {108,108,108}};
+
+const size_t WebRtcIlbcfix_kFilterRange[5]={63, 85, 125, 147, 147};
+
+/* Gain Quantization for the codebook gains of the 3 stages */
+
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq3[9]={
+  -16384, -10813, -5407, 0, 4096, 8192,
+  12288, 16384, 32767};
+
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq4[17]={
+  -17203, -14746, -12288, -9830, -7373, -4915,
+  -2458, 0, 2458, 4915, 7373, 9830,
+  12288, 14746, 17203, 19661, 32767};
+
+/* Q14 (one extra value (max int16_t) to simplify for the search) */
+const int16_t WebRtcIlbcfix_kGainSq5[33]={
+  614,        1229,        1843,        2458,        3072,       3686,
+  4301,        4915,        5530,        6144,        6758,        7373,
+  7987,        8602,        9216,        9830,       10445,       11059,
+  11674,       12288,       12902,       13517,       14131,       14746,
+  15360,       15974,       16589,       17203,       17818,       18432,
+  19046,       19661,    32767};
+
+/* Q14 gain_sq5Tbl squared in Q14 */
+const int16_t WebRtcIlbcfix_kGainSq5Sq[32] = {
+  23,   92,    207,  368,  576,  829,
+  1129,  1474,   1866,  2304,  2787,  3317,
+  3893,  4516,   5184,  5897,  6658,  7464,
+  8318,  9216,   10160,  11151,  12187,  13271,
+  14400,  15574,   16796,  18062,  19377,  20736,
+  22140,  23593
+};
+
+const int16_t* const WebRtcIlbcfix_kGain[3] =
+{WebRtcIlbcfix_kGainSq5, WebRtcIlbcfix_kGainSq4, WebRtcIlbcfix_kGainSq3};
+
+
+/* Tables for the Enhancer, using upsamling factor 4 (ENH_UPS0 = 4) */
+
+const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1]={
+  {0,    0,    0, 4096,    0,  0,   0},
+  {64, -315, 1181, 3531, -436, 77, -64},
+  {97, -509, 2464, 2464, -509, 97, -97},
+  {77, -436, 3531, 1181, -315, 64, -77}
+};
+
+const int16_t WebRtcIlbcfix_kEnhWt[3] = {
+  4800, 16384, 27968 /* Q16 */
+};
+
+const size_t WebRtcIlbcfix_kEnhPlocs[ENH_NBLOCKS_TOT] = {
+  160, 480, 800, 1120, 1440, 1760, 2080, 2400  /* Q(-2) */
+};
+
+/* PLC table */
+
+const int16_t WebRtcIlbcfix_kPlcPerSqr[6] = { /* Grid points for square of periodiciy in Q15 */
+  839, 1343, 2048, 2998, 4247, 5849
+};
+
+const int16_t WebRtcIlbcfix_kPlcPitchFact[6] = { /* Value of y=(x^4-0.4)/(0.7-0.4) in grid points in Q15 */
+  0, 5462, 10922, 16384, 21846, 27306
+};
+
+const int16_t WebRtcIlbcfix_kPlcPfSlope[6] = { /* Slope of y=(x^4-0.4)/(0.7-0.4) in Q11 */
+  26667, 18729, 13653, 10258, 7901, 6214
+};
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.h
new file mode 100644
index 0000000000..a8645c00db
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/constants.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ constants.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* high pass filters */
+
+extern const int16_t WebRtcIlbcfix_kHpInCoefs[];
+extern const int16_t WebRtcIlbcfix_kHpOutCoefs[];
+
+/* Window for start state decision */
+extern const int16_t WebRtcIlbcfix_kStartSequenceEnrgWin[];
+
+/* low pass filter used for downsampling */
+extern const int16_t WebRtcIlbcfix_kLpFiltCoefs[];
+
+/* LPC analysis and quantization */
+
+extern const int16_t WebRtcIlbcfix_kLpcWin[];
+extern const int16_t WebRtcIlbcfix_kLpcAsymWin[];
+extern const int32_t WebRtcIlbcfix_kLpcLagWin[];
+extern const int16_t WebRtcIlbcfix_kLpcChirpSyntDenum[];
+extern const int16_t WebRtcIlbcfix_kLpcChirpWeightDenum[];
+extern const int16_t WebRtcIlbcfix_kLsfDimCb[];
+extern const int16_t WebRtcIlbcfix_kLsfSizeCb[];
+extern const int16_t WebRtcIlbcfix_kLsfCb[];
+extern const int16_t WebRtcIlbcfix_kLsfWeight20ms[];
+extern const int16_t WebRtcIlbcfix_kLsfWeight30ms[];
+extern const int16_t WebRtcIlbcfix_kLsfMean[];
+extern const int16_t WebRtcIlbcfix_kLspMean[];
+extern const int16_t WebRtcIlbcfix_kCos[];
+extern const int16_t WebRtcIlbcfix_kCosDerivative[];
+extern const int16_t WebRtcIlbcfix_kCosGrid[];
+extern const int16_t WebRtcIlbcfix_kAcosDerivative[];
+
+/* state quantization tables */
+
+extern const int16_t WebRtcIlbcfix_kStateSq3[];
+extern const int32_t WebRtcIlbcfix_kChooseFrgQuant[];
+extern const int16_t WebRtcIlbcfix_kScale[];
+extern const int16_t WebRtcIlbcfix_kFrgQuantMod[];
+
+/* Ranges for search and filters at different subframes */
+
+extern const size_t WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES];
+extern const size_t WebRtcIlbcfix_kFilterRange[];
+
+/* gain quantization tables */
+
+extern const int16_t WebRtcIlbcfix_kGainSq3[];
+extern const int16_t WebRtcIlbcfix_kGainSq4[];
+extern const int16_t WebRtcIlbcfix_kGainSq5[];
+extern const int16_t WebRtcIlbcfix_kGainSq5Sq[];
+extern const int16_t* const WebRtcIlbcfix_kGain[];
+
+/* adaptive codebook definitions */
+
+extern const int16_t WebRtcIlbcfix_kCbFiltersRev[];
+extern const int16_t WebRtcIlbcfix_kAlpha[];
+
+/* enhancer definitions */
+
+extern const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0]
+                                                 [ENH_FLO_MULT2_PLUS1];
+extern const int16_t WebRtcIlbcfix_kEnhWt[];
+extern const size_t WebRtcIlbcfix_kEnhPlocs[];
+
+/* PLC tables */
+
+extern const int16_t WebRtcIlbcfix_kPlcPerSqr[];
+extern const int16_t WebRtcIlbcfix_kPlcPitchFact[];
+extern const int16_t WebRtcIlbcfix_kPlcPfSlope[];
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
new file mode 100644
index 0000000000..7e21faee6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CreateAugmentedVec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "rtc_base/sanitizer.h"
+
+/*----------------------------------------------------------------*
+ *  Recreate a specific codebook vector from the augmented part.
+ *
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CreateAugmentedVec(
+    size_t index,          /* (i) Index for the augmented vector to be
+                              created */
+    const int16_t* buffer, /* (i) Pointer to the end of the codebook memory
+                              that is used for creation of the augmented
+                              codebook */
+    int16_t* cbVec) {      /* (o) The constructed codebook vector */
+  size_t ilow;
+  const int16_t *ppo, *ppi;
+  int16_t cbVecTmp[4];
+  /* Interpolation starts 4 elements before cbVec+index, but must not start
+     outside `cbVec`; clamping interp_len to stay within `cbVec`.
+   */
+  size_t interp_len = WEBRTC_SPL_MIN(index, 4);
+
+  rtc_MsanCheckInitialized(buffer - index - interp_len, sizeof(buffer[0]),
+                           index + interp_len);
+
+  ilow = index - interp_len;
+
+  /* copy the first noninterpolated part */
+  ppo = buffer-index;
+  WEBRTC_SPL_MEMCPY_W16(cbVec, ppo, index);
+
+  /* interpolation */
+  ppo = buffer - interp_len;
+  ppi = buffer - index - interp_len;
+
+  /* perform cbVec[ilow+k] = ((ppi[k]*alphaTbl[k])>>15) +
+                             ((ppo[k]*alphaTbl[interp_len-1-k])>>15);
+     for k = 0..interp_len-1
+  */
+  WebRtcSpl_ElementwiseVectorMult(&cbVec[ilow], ppi, WebRtcIlbcfix_kAlpha,
+                                  interp_len, 15);
+  WebRtcSpl_ReverseOrderMultArrayElements(
+      cbVecTmp, ppo, &WebRtcIlbcfix_kAlpha[interp_len - 1], interp_len, 15);
+  WebRtcSpl_AddVectorsAndShift(&cbVec[ilow], &cbVec[ilow], cbVecTmp, interp_len,
+                               0);
+
+  /* copy the second noninterpolated part */
+  ppo = buffer - index;
+  /* `tempbuff2` is declared in WebRtcIlbcfix_GetCbVec and is SUBL+5 elements
+     long. `buffer` points one element past the end of that vector, i.e., at
+     tempbuff2+SUBL+5. Since ppo=buffer-index, we cannot read any more than
+     `index` elements from `ppo`.
+
+     `cbVec` is declared to be SUBL elements long in WebRtcIlbcfix_CbConstruct.
+     Therefore, we can only write SUBL-index elements to cbVec+index.
+
+     These two conditions limit the number of elements to copy.
+   */
+  WEBRTC_SPL_MEMCPY_W16(cbVec+index, ppo, WEBRTC_SPL_MIN(SUBL-index, index));
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
new file mode 100644
index 0000000000..d7e5be1c2f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_CreateAugmentedVec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Recreate a specific codebook vector from the augmented part.
+ *
+ *----------------------------------------------------------------*/
+
+void WebRtcIlbcfix_CreateAugmentedVec(
+    size_t index, /* (i) Index for the augmented vector to be
+                     created */
+    const int16_t* buffer, /* (i) Pointer to the end of the codebook memory
+                              that is used for creation of the augmented
+                              codebook */
+    int16_t* cbVec);       /* (o) The construced codebook vector */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.c
new file mode 100644
index 0000000000..d7621d5b65
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.c
@@ -0,0 +1,261 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Decode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/decode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/decode_residual.h"
+#include "modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h"
+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+#include "rtc_base/system/arch.h"
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
+#endif
+
+/*----------------------------------------------------------------*
+ *  main decoder function
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_DecodeImpl(
+    int16_t *decblock,    /* (o) decoded signal block */
+    const uint16_t *bytes, /* (i) encoded signal bits */
+    IlbcDecoder *iLBCdec_inst, /* (i/o) the decoder state
+                                           structure */
+    int16_t mode      /* (i) 0: bad packet, PLC,
+                                                                   1: normal */
+                           ) {
+  const int old_mode = iLBCdec_inst->mode;
+  const int old_use_enhancer = iLBCdec_inst->use_enhancer;
+
+  size_t i;
+  int16_t order_plus_one;
+
+  int16_t last_bit;
+  int16_t *data;
+  /* Stack based */
+  int16_t decresidual[BLOCKL_MAX];
+  int16_t PLCresidual[BLOCKL_MAX + LPC_FILTERORDER];
+  int16_t syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
+  int16_t PLClpc[LPC_FILTERORDER + 1];
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+  uint16_t swapped[NO_OF_WORDS_30MS];
+#endif
+  iLBC_bits *iLBCbits_inst = (iLBC_bits*)PLCresidual;
+
+  /* Reuse some buffers that are non overlapping in order to save stack memory */
+  data = &PLCresidual[LPC_FILTERORDER];
+
+  if (mode) { /* the data are good */
+
+    /* decode data */
+
+    /* Unpacketize bits into parameters */
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+    WebRtcIlbcfix_SwapBytes(bytes, iLBCdec_inst->no_of_words, swapped);
+    last_bit = WebRtcIlbcfix_UnpackBits(swapped, iLBCbits_inst, iLBCdec_inst->mode);
+#else
+    last_bit = WebRtcIlbcfix_UnpackBits(bytes, iLBCbits_inst, iLBCdec_inst->mode);
+#endif
+
+    /* Check for bit errors */
+    if (iLBCbits_inst->startIdx<1)
+      mode = 0;
+    if ((iLBCdec_inst->mode==20) && (iLBCbits_inst->startIdx>3))
+      mode = 0;
+    if ((iLBCdec_inst->mode==30) && (iLBCbits_inst->startIdx>5))
+      mode = 0;
+    if (last_bit==1)
+      mode = 0;
+
+    if (mode) { /* No bit errors was detected, continue decoding */
+      /* Stack based */
+      int16_t lsfdeq[LPC_FILTERORDER*LPC_N_MAX];
+      int16_t weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+
+      /* adjust index */
+      WebRtcIlbcfix_IndexConvDec(iLBCbits_inst->cb_index);
+
+      /* decode the lsf */
+      WebRtcIlbcfix_SimpleLsfDeQ(lsfdeq, (int16_t*)(iLBCbits_inst->lsf), iLBCdec_inst->lpc_n);
+      WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCdec_inst->lpc_n);
+      WebRtcIlbcfix_DecoderInterpolateLsp(syntdenum, weightdenum,
+                                          lsfdeq, LPC_FILTERORDER, iLBCdec_inst);
+
+      /* Decode the residual using the cb and gain indexes */
+      if (!WebRtcIlbcfix_DecodeResidual(iLBCdec_inst, iLBCbits_inst,
+                                        decresidual, syntdenum))
+        goto error;
+
+      /* preparing the plc for a future loss! */
+      WebRtcIlbcfix_DoThePlc(
+          PLCresidual, PLClpc, 0, decresidual,
+          syntdenum + (LPC_FILTERORDER + 1) * (iLBCdec_inst->nsub - 1),
+          iLBCdec_inst->last_lag, iLBCdec_inst);
+
+      /* Use the output from doThePLC */
+      WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
+    }
+
+  }
+
+  if (mode == 0) {
+    /* the data is bad (either a PLC call
+     * was made or a bit error was detected)
+     */
+
+    /* packet loss conceal */
+
+    WebRtcIlbcfix_DoThePlc(PLCresidual, PLClpc, 1, decresidual, syntdenum,
+                           iLBCdec_inst->last_lag, iLBCdec_inst);
+
+    WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
+
+    order_plus_one = LPC_FILTERORDER + 1;
+
+    for (i = 0; i < iLBCdec_inst->nsub; i++) {
+      WEBRTC_SPL_MEMCPY_W16(syntdenum+(i*order_plus_one),
+                            PLClpc, order_plus_one);
+    }
+  }
+
+  if ((*iLBCdec_inst).use_enhancer == 1) { /* Enhancer activated */
+
+    /* Update the filter and filter coefficients if there was a packet loss */
+    if (iLBCdec_inst->prev_enh_pl==2) {
+      for (i=0;i<iLBCdec_inst->nsub;i++) {
+        WEBRTC_SPL_MEMCPY_W16(&(iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)]),
+                              syntdenum, (LPC_FILTERORDER+1));
+      }
+    }
+
+    /* post filtering */
+    (*iLBCdec_inst).last_lag =
+        WebRtcIlbcfix_EnhancerInterface(data, decresidual, iLBCdec_inst);
+
+    /* synthesis filtering */
+
+    /* Set up the filter state */
+    WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER);
+
+    if (iLBCdec_inst->mode==20) {
+      /* Enhancer has 40 samples delay */
+      i=0;
+      WebRtcSpl_FilterARFastQ12(
+          data, data,
+          iLBCdec_inst->old_syntdenum + (i+iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1),
+          LPC_FILTERORDER+1, SUBL);
+
+      for (i=1; i < iLBCdec_inst->nsub; i++) {
+        WebRtcSpl_FilterARFastQ12(
+            data+i*SUBL, data+i*SUBL,
+            syntdenum+(i-1)*(LPC_FILTERORDER+1),
+            LPC_FILTERORDER+1, SUBL);
+      }
+
+    } else if (iLBCdec_inst->mode==30) {
+      /* Enhancer has 80 samples delay */
+      for (i=0; i < 2; i++) {
+        WebRtcSpl_FilterARFastQ12(
+            data+i*SUBL, data+i*SUBL,
+            iLBCdec_inst->old_syntdenum + (i+4)*(LPC_FILTERORDER+1),
+            LPC_FILTERORDER+1, SUBL);
+      }
+      for (i=2; i < iLBCdec_inst->nsub; i++) {
+        WebRtcSpl_FilterARFastQ12(
+            data+i*SUBL, data+i*SUBL,
+            syntdenum+(i-2)*(LPC_FILTERORDER+1),
+            LPC_FILTERORDER+1, SUBL);
+      }
+    }
+
+    /* Save the filter state */
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+
+  } else { /* Enhancer not activated */
+    size_t lag;
+
+    /* Find last lag (since the enhancer is not called to give this info) */
+    lag = 20;
+    if (iLBCdec_inst->mode==20) {
+      lag = WebRtcIlbcfix_XcorrCoef(
+          &decresidual[iLBCdec_inst->blockl-60],
+          &decresidual[iLBCdec_inst->blockl-60-lag],
+          60,
+          80, lag, -1);
+    } else {
+      lag = WebRtcIlbcfix_XcorrCoef(
+          &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL],
+          &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL-lag],
+          ENH_BLOCKL,
+          100, lag, -1);
+    }
+
+    /* Store lag (it is needed if next packet is lost) */
+    (*iLBCdec_inst).last_lag = lag;
+
+    /* copy data and run synthesis filter */
+    WEBRTC_SPL_MEMCPY_W16(data, decresidual, iLBCdec_inst->blockl);
+
+    /* Set up the filter state */
+    WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER);
+
+    for (i=0; i < iLBCdec_inst->nsub; i++) {
+      WebRtcSpl_FilterARFastQ12(
+          data+i*SUBL, data+i*SUBL,
+          syntdenum + i*(LPC_FILTERORDER+1),
+          LPC_FILTERORDER+1, SUBL);
+    }
+
+    /* Save the filter state */
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+  }
+
+  WEBRTC_SPL_MEMCPY_W16(decblock,data,iLBCdec_inst->blockl);
+
+  /* High pass filter the signal (with upscaling a factor 2 and saturation) */
+  WebRtcIlbcfix_HpOutput(decblock, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
+                         iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+                         iLBCdec_inst->blockl);
+
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->old_syntdenum,
+                        syntdenum, iLBCdec_inst->nsub*(LPC_FILTERORDER+1));
+
+  iLBCdec_inst->prev_enh_pl=0;
+
+  if (mode==0) { /* PLC was used */
+    iLBCdec_inst->prev_enh_pl=1;
+  }
+
+  return 0;  // Success.
+
+error:
+  // The decoder got sick from eating that data. Reset it and return.
+  WebRtcIlbcfix_InitDecode(iLBCdec_inst, old_mode, old_use_enhancer);
+  return -1;  // Error
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.h
new file mode 100644
index 0000000000..a7d2910115
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode.h
@@ -0,0 +1,42 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Decode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
+
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  main decoder function
+ *---------------------------------------------------------------*/
+
+// Returns 0 on success, -1 on error.
+ABSL_MUST_USE_RESULT
+int WebRtcIlbcfix_DecodeImpl(
+    int16_t* decblock,         /* (o) decoded signal block */
+    const uint16_t* bytes,     /* (i) encoded signal bits */
+    IlbcDecoder* iLBCdec_inst, /* (i/o) the decoder state
+                                           structure */
+    int16_t mode               /* (i) 0: bad packet, PLC,
+                                      1: normal */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
new file mode 100644
index 0000000000..a9668e2889
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecodeResidual.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/decode_residual.h"
+
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+
+/*----------------------------------------------------------------*
+ *  frame residual decoder function (subrutine to iLBC_decode)
+ *---------------------------------------------------------------*/
+
+bool WebRtcIlbcfix_DecodeResidual(
+    IlbcDecoder *iLBCdec_inst,
+    /* (i/o) the decoder state structure */
+    iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
+                                for the decoding  */
+    int16_t *decresidual,  /* (o) decoded residual frame */
+    int16_t *syntdenum   /* (i) the decoded synthesis filter
+                                  coefficients */
+                                  ) {
+  size_t meml_gotten, diff, start_pos;
+  size_t subcount, subframe;
+  int16_t *reverseDecresidual = iLBCdec_inst->enh_buf; /* Reversed decoded data, used for decoding backwards in time (reuse memory in state) */
+  int16_t *memVec = iLBCdec_inst->prevResidual;  /* Memory for codebook and filter state (reuse memory in state) */
+  int16_t *mem = &memVec[CB_HALFFILTERLEN];   /* Memory for codebook */
+
+  diff = STATE_LEN - iLBCdec_inst->state_short_len;
+
+  if (iLBC_encbits->state_first == 1) {
+    start_pos = (iLBC_encbits->startIdx-1)*SUBL;
+  } else {
+    start_pos = (iLBC_encbits->startIdx-1)*SUBL + diff;
+  }
+
+  /* decode scalar part of start state */
+
+  WebRtcIlbcfix_StateConstruct(iLBC_encbits->idxForMax,
+                               iLBC_encbits->idxVec, &syntdenum[(iLBC_encbits->startIdx-1)*(LPC_FILTERORDER+1)],
+                               &decresidual[start_pos], iLBCdec_inst->state_short_len
+                               );
+
+  if (iLBC_encbits->state_first) { /* put adaptive part in the end */
+
+    /* setup memory */
+
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCdec_inst->state_short_len);
+    WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCdec_inst->state_short_len, decresidual+start_pos,
+                          iLBCdec_inst->state_short_len);
+
+    /* construct decoded vector */
+
+    if (!WebRtcIlbcfix_CbConstruct(
+            &decresidual[start_pos + iLBCdec_inst->state_short_len],
+            iLBC_encbits->cb_index, iLBC_encbits->gain_index,
+            mem + CB_MEML - ST_MEM_L_TBL, ST_MEM_L_TBL, diff))
+      return false;  // Error.
+
+  }
+  else {/* put adaptive part in the beginning */
+
+    /* setup memory */
+
+    meml_gotten = iLBCdec_inst->state_short_len;
+    WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
+                                  decresidual+start_pos, meml_gotten);
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
+
+    /* construct decoded vector */
+
+    if (!WebRtcIlbcfix_CbConstruct(reverseDecresidual, iLBC_encbits->cb_index,
+                                   iLBC_encbits->gain_index,
+                                   mem + CB_MEML - ST_MEM_L_TBL, ST_MEM_L_TBL,
+                                   diff))
+      return false;  // Error.
+
+    /* get decoded residual from reversed vector */
+
+    WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1],
+                                  reverseDecresidual, diff);
+  }
+
+  /* counter for predicted subframes */
+
+  subcount=1;
+
+  /* forward prediction of subframes */
+
+  if (iLBCdec_inst->nsub > iLBC_encbits->startIdx + 1) {
+
+    /* setup memory */
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
+    WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN,
+                          decresidual+(iLBC_encbits->startIdx-1)*SUBL, STATE_LEN);
+
+    /* loop over subframes to encode */
+
+    size_t Nfor = iLBCdec_inst->nsub - iLBC_encbits->startIdx - 1;
+    for (subframe=0; subframe<Nfor; subframe++) {
+
+      /* construct decoded vector */
+      if (!WebRtcIlbcfix_CbConstruct(
+              &decresidual[(iLBC_encbits->startIdx + 1 + subframe) * SUBL],
+              iLBC_encbits->cb_index + subcount * CB_NSTAGES,
+              iLBC_encbits->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+              SUBL))
+        return false;  // Error;
+
+      /* update memory */
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &decresidual[(iLBC_encbits->startIdx+1+subframe)*SUBL], SUBL);
+
+      subcount++;
+    }
+
+  }
+
+  /* backward prediction of subframes */
+
+  if (iLBC_encbits->startIdx > 1) {
+
+    /* setup memory */
+
+    meml_gotten = SUBL*(iLBCdec_inst->nsub+1-iLBC_encbits->startIdx);
+    if( meml_gotten > CB_MEML ) {
+      meml_gotten=CB_MEML;
+    }
+
+    WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
+                                  decresidual+(iLBC_encbits->startIdx-1)*SUBL, meml_gotten);
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
+
+    /* loop over subframes to decode */
+
+    size_t Nback = iLBC_encbits->startIdx - 1;
+    for (subframe=0; subframe<Nback; subframe++) {
+
+      /* construct decoded vector */
+      if (!WebRtcIlbcfix_CbConstruct(
+              &reverseDecresidual[subframe * SUBL],
+              iLBC_encbits->cb_index + subcount * CB_NSTAGES,
+              iLBC_encbits->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+              SUBL))
+        return false;  // Error.
+
+      /* update memory */
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &reverseDecresidual[subframe*SUBL], SUBL);
+
+      subcount++;
+    }
+
+    /* get decoded residual from reversed vector */
+    WebRtcSpl_MemCpyReversedOrder(decresidual+SUBL*Nback-1,
+                                  reverseDecresidual, SUBL*Nback);
+  }
+
+  return true;  // Success.
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
new file mode 100644
index 0000000000..d079577661
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecodeResidual.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
+
+#include <stdbool.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  frame residual decoder function (subrutine to iLBC_decode)
+ *---------------------------------------------------------------*/
+
+// Returns true on success, false on failure. In case of failure, the decoder
+// state may be corrupted and needs resetting.
+ABSL_MUST_USE_RESULT
+bool WebRtcIlbcfix_DecodeResidual(
+    IlbcDecoder* iLBCdec_inst, /* (i/o) the decoder state structure */
+    iLBC_bits* iLBC_encbits,   /* (i/o) Encoded bits, which are used
+                                        for the decoding  */
+    int16_t* decresidual,      /* (o) decoded residual frame */
+    int16_t* syntdenum         /* (i) the decoded synthesis filter
+                                                         coefficients */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
new file mode 100644
index 0000000000..d96bb9b2e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecoderInterpolateLsp.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h"
+
+/*----------------------------------------------------------------*
+ *  obtain synthesis and weighting filters form lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecoderInterpolateLsp(
+    int16_t *syntdenum,  /* (o) synthesis filter coefficients */
+    int16_t *weightdenum, /* (o) weighting denumerator
+                                   coefficients */
+    int16_t *lsfdeq,   /* (i) dequantized lsf coefficients */
+    int16_t length,   /* (i) length of lsf coefficient vector */
+    IlbcDecoder *iLBCdec_inst
+    /* (i) the decoder state structure */
+                                          ){
+  size_t i;
+  int pos, lp_length;
+  int16_t  lp[LPC_FILTERORDER + 1], *lsfdeq2;
+
+  lsfdeq2 = lsfdeq + length;
+  lp_length = length + 1;
+
+  if (iLBCdec_inst->mode==30) {
+    /* subframe 1: Interpolation between old and first LSF */
+
+    WebRtcIlbcfix_LspInterpolate2PolyDec(lp, (*iLBCdec_inst).lsfdeqold, lsfdeq,
+                                         WebRtcIlbcfix_kLsfWeight30ms[0], length);
+    WEBRTC_SPL_MEMCPY_W16(syntdenum,lp,lp_length);
+    WebRtcIlbcfix_BwExpand(weightdenum, lp, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
+
+    /* subframes 2 to 6: interpolation between first and last LSF */
+
+    pos = lp_length;
+    for (i = 1; i < 6; i++) {
+      WebRtcIlbcfix_LspInterpolate2PolyDec(lp, lsfdeq, lsfdeq2,
+                                           WebRtcIlbcfix_kLsfWeight30ms[i], length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum + pos,lp,lp_length);
+      WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
+      pos += lp_length;
+    }
+  } else { /* iLBCdec_inst->mode=20 */
+    /* subframes 1 to 4: interpolation between old and new LSF */
+    pos = 0;
+    for (i = 0; i < iLBCdec_inst->nsub; i++) {
+      WebRtcIlbcfix_LspInterpolate2PolyDec(lp, iLBCdec_inst->lsfdeqold, lsfdeq,
+                                           WebRtcIlbcfix_kLsfWeight20ms[i], length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum+pos,lp,lp_length);
+      WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, (int16_t)lp_length);
+      pos += lp_length;
+    }
+  }
+
+  /* update memory */
+
+  if (iLBCdec_inst->mode==30) {
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq2, length);
+  } else {
+    WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq, length);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
new file mode 100644
index 0000000000..8b08114467
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DecoderInterpolateLsp.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  obtain synthesis and weighting filters form lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DecoderInterpolateLsp(
+    int16_t* syntdenum, /* (o) synthesis filter coefficients */
+    int16_t* weightdenum, /* (o) weighting denumerator
+                                   coefficients */
+    int16_t* lsfdeq, /* (i) dequantized lsf coefficients */
+    int16_t length,  /* (i) length of lsf coefficient vector */
+    IlbcDecoder* iLBCdec_inst
+    /* (i) the decoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/defines.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/defines.h
new file mode 100644
index 0000000000..64135c4887
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/defines.h
@@ -0,0 +1,225 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ define.h
+
+******************************************************************/
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
+
+#include <stdint.h>
+#include <string.h>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+/* general codec settings */
+
+#define FS 8000
+#define BLOCKL_20MS 160
+#define BLOCKL_30MS 240
+#define BLOCKL_MAX 240
+#define NSUB_20MS 4
+#define NSUB_30MS 6
+#define NSUB_MAX 6
+#define NASUB_20MS 2
+#define NASUB_30MS 4
+#define NASUB_MAX 4
+#define SUBL 40
+#define STATE_LEN 80
+#define STATE_SHORT_LEN_30MS 58
+#define STATE_SHORT_LEN_20MS 57
+
+/* LPC settings */
+
+#define LPC_FILTERORDER 10
+#define LPC_LOOKBACK 60
+#define LPC_N_20MS 1
+#define LPC_N_30MS 2
+#define LPC_N_MAX 2
+#define LPC_ASYMDIFF 20
+#define LSF_NSPLIT 3
+#define LSF_NUMBER_OF_STEPS 4
+#define LPC_HALFORDER 5
+#define COS_GRID_POINTS 60
+
+/* cb settings */
+
+#define CB_NSTAGES 3
+#define CB_EXPAND 2
+#define CB_MEML 147
+#define CB_FILTERLEN (2 * 4)
+#define CB_HALFFILTERLEN 4
+#define CB_RESRANGE 34
+#define CB_MAXGAIN_FIXQ6 83 /* error = -0.24% */
+#define CB_MAXGAIN_FIXQ14 21299
+
+/* enhancer */
+
+#define ENH_BLOCKL 80 /* block length */
+#define ENH_BLOCKL_HALF (ENH_BLOCKL / 2)
+#define ENH_HL                                                         \
+  3 /* 2*ENH_HL+1 is number blocks                                     \
+                                                        in said second \
+       sequence */
+#define ENH_SLOP                    \
+  2 /* max difference estimated and \
+                                                       correct pitch period */
+#define ENH_PLOCSL                                                          \
+  8 /* pitch-estimates and                                                  \
+                                                     pitch-locations buffer \
+       length */
+#define ENH_OVERHANG 2
+#define ENH_UPS0 4 /* upsampling rate */
+#define ENH_FL0 3  /* 2*FLO+1 is the length of each filter */
+#define ENH_FLO_MULT2_PLUS1 7
+#define ENH_VECTL (ENH_BLOCKL + 2 * ENH_FL0)
+#define ENH_CORRDIM (2 * ENH_SLOP + 1)
+#define ENH_NBLOCKS (BLOCKL / ENH_BLOCKL)
+#define ENH_NBLOCKS_EXTRA 5
+#define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS+ENH_NBLOCKS_EXTRA */
+#define ENH_BUFL (ENH_NBLOCKS_TOT) * ENH_BLOCKL
+#define ENH_BUFL_FILTEROVERHEAD 3
+#define ENH_A0 819                      /* Q14 */
+#define ENH_A0_MINUS_A0A0DIV4 848256041 /* Q34 */
+#define ENH_A0DIV2 26843546             /* Q30 */
+
+/* PLC */
+
+/* Down sampling */
+
+#define FILTERORDER_DS_PLUS1 7
+#define DELAY_DS 3
+#define FACTOR_DS 2
+
+/* bit stream defs */
+
+#define NO_OF_BYTES_20MS 38
+#define NO_OF_BYTES_30MS 50
+#define NO_OF_WORDS_20MS 19
+#define NO_OF_WORDS_30MS 25
+#define STATE_BITS 3
+#define BYTE_LEN 8
+#define ULP_CLASSES 3
+
+/* help parameters */
+
+#define TWO_PI_FIX 25736 /* Q12 */
+
+/* Constants for codebook search and creation */
+
+#define ST_MEM_L_TBL 85
+#define MEM_LF_TBL 147
+
+/* Struct for the bits */
+typedef struct iLBC_bits_t_ {
+  int16_t lsf[LSF_NSPLIT * LPC_N_MAX];
+  int16_t cb_index[CB_NSTAGES * (NASUB_MAX + 1)];   /* First CB_NSTAGES values
+                                                       contains extra CB index */
+  int16_t gain_index[CB_NSTAGES * (NASUB_MAX + 1)]; /* First CB_NSTAGES values
+                                                       contains extra CB gain */
+  size_t idxForMax;
+  int16_t state_first;
+  int16_t idxVec[STATE_SHORT_LEN_30MS];
+  int16_t firstbits;
+  size_t startIdx;
+} iLBC_bits;
+
+/* type definition encoder instance */
+typedef struct IlbcEncoder_ {
+  /* flag for frame size mode */
+  int16_t mode;
+
+  /* basic parameters for different frame sizes */
+  size_t blockl;
+  size_t nsub;
+  int16_t nasub;
+  size_t no_of_bytes, no_of_words;
+  int16_t lpc_n;
+  size_t state_short_len;
+
+  /* analysis filter state */
+  int16_t anaMem[LPC_FILTERORDER];
+
+  /* Fix-point old lsf parameters for interpolation */
+  int16_t lsfold[LPC_FILTERORDER];
+  int16_t lsfdeqold[LPC_FILTERORDER];
+
+  /* signal buffer for LP analysis */
+  int16_t lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
+
+  /* state of input HP filter */
+  int16_t hpimemx[2];
+  int16_t hpimemy[4];
+
+#ifdef SPLIT_10MS
+  int16_t weightdenumbuf[66];
+  int16_t past_samples[160];
+  uint16_t bytes[25];
+  int16_t section;
+  int16_t Nfor_flag;
+  int16_t Nback_flag;
+  int16_t start_pos;
+  size_t diff;
+#endif
+
+} IlbcEncoder;
+
+/* type definition decoder instance */
+typedef struct IlbcDecoder_ {
+  /* flag for frame size mode */
+  int16_t mode;
+
+  /* basic parameters for different frame sizes */
+  size_t blockl;
+  size_t nsub;
+  int16_t nasub;
+  size_t no_of_bytes, no_of_words;
+  int16_t lpc_n;
+  size_t state_short_len;
+
+  /* synthesis filter state */
+  int16_t syntMem[LPC_FILTERORDER];
+
+  /* old LSF for interpolation */
+  int16_t lsfdeqold[LPC_FILTERORDER];
+
+  /* pitch lag estimated in enhancer and used in PLC */
+  size_t last_lag;
+
+  /* PLC state information */
+  int consPLICount, prev_enh_pl;
+  int16_t perSquare;
+
+  int16_t prevScale, prevPLI;
+  size_t prevLag;
+  int16_t prevLpc[LPC_FILTERORDER + 1];
+  int16_t prevResidual[NSUB_MAX * SUBL];
+  int16_t seed;
+
+  /* previous synthesis filter parameters */
+
+  int16_t old_syntdenum[(LPC_FILTERORDER + 1) * NSUB_MAX];
+
+  /* state of output HP filter */
+  int16_t hpimemx[2];
+  int16_t hpimemy[4];
+
+  /* enhancer state information */
+  int use_enhancer;
+  int16_t enh_buf[ENH_BUFL + ENH_BUFL_FILTEROVERHEAD];
+  size_t enh_period[ENH_NBLOCKS_TOT];
+
+} IlbcDecoder;
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.c
new file mode 100644
index 0000000000..9ca6ca48e9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.c
@@ -0,0 +1,309 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DoThePlc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/comp_corr.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Packet loss concealment routine. Conceals a residual signal
+ *  and LP parameters. If no packet loss, update state.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DoThePlc(
+    int16_t *PLCresidual,  /* (o) concealed residual */
+    int16_t *PLClpc,    /* (o) concealed LP parameters */
+    int16_t PLI,     /* (i) packet loss indicator
+                                                           0 - no PL, 1 = PL */
+    int16_t *decresidual,  /* (i) decoded residual */
+    int16_t *lpc,    /* (i) decoded LPC (only used for no PL) */
+    size_t inlag,    /* (i) pitch lag */
+    IlbcDecoder *iLBCdec_inst
+    /* (i/o) decoder instance */
+                            ){
+  size_t i;
+  int32_t cross, ener, cross_comp, ener_comp = 0;
+  int32_t measure, maxMeasure, energy;
+  int32_t noise_energy_threshold_30dB;
+  int16_t max, crossSquareMax, crossSquare;
+  size_t j, lag, randlag;
+  int16_t tmp1, tmp2;
+  int16_t shift1, shift2, shift3, shiftMax;
+  int16_t scale3;
+  size_t corrLen;
+  int32_t tmpW32, tmp2W32;
+  int16_t use_gain;
+  int16_t tot_gain;
+  int16_t max_perSquare;
+  int16_t scale1, scale2;
+  int16_t totscale;
+  int32_t nom;
+  int16_t denom;
+  int16_t pitchfact;
+  size_t use_lag;
+  int ind;
+  int16_t randvec[BLOCKL_MAX];
+
+  /* Packet Loss */
+  if (PLI == 1) {
+
+    (*iLBCdec_inst).consPLICount += 1;
+
+    /* if previous frame not lost,
+       determine pitch pred. gain */
+
+    if (iLBCdec_inst->prevPLI != 1) {
+
+      /* Maximum 60 samples are correlated, preserve as high accuracy
+         as possible without getting overflow */
+      max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual,
+                                     iLBCdec_inst->blockl);
+      scale3 = (WebRtcSpl_GetSizeInBits(max)<<1) - 25;
+      if (scale3 < 0) {
+        scale3 = 0;
+      }
+
+      /* Store scale for use when interpolating between the
+       * concealment and the received packet */
+      iLBCdec_inst->prevScale = scale3;
+
+      /* Search around the previous lag +/-3 to find the
+         best pitch period */
+      lag = inlag - 3;
+
+      /* Guard against getting outside the frame */
+      corrLen = (size_t)WEBRTC_SPL_MIN(60, iLBCdec_inst->blockl-(inlag+3));
+
+      WebRtcIlbcfix_CompCorr( &cross, &ener,
+                              iLBCdec_inst->prevResidual, lag, iLBCdec_inst->blockl, corrLen, scale3);
+
+      /* Normalize and store cross^2 and the number of shifts */
+      shiftMax = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross))-15;
+      crossSquareMax = (int16_t)((
+          (int16_t)WEBRTC_SPL_SHIFT_W32(cross, -shiftMax) *
+          (int16_t)WEBRTC_SPL_SHIFT_W32(cross, -shiftMax)) >> 15);
+
+      for (j=inlag-2;j<=inlag+3;j++) {
+        WebRtcIlbcfix_CompCorr( &cross_comp, &ener_comp,
+                                iLBCdec_inst->prevResidual, j, iLBCdec_inst->blockl, corrLen, scale3);
+
+        /* Use the criteria (corr*corr)/energy to compare if
+           this lag is better or not. To avoid the division,
+           do a cross multiplication */
+        shift1 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross_comp))-15;
+        crossSquare = (int16_t)((
+            (int16_t)WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1) *
+            (int16_t)WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1)) >> 15);
+
+        shift2 = WebRtcSpl_GetSizeInBits(ener)-15;
+        measure = (int16_t)WEBRTC_SPL_SHIFT_W32(ener, -shift2) * crossSquare;
+
+        shift3 = WebRtcSpl_GetSizeInBits(ener_comp)-15;
+        maxMeasure = (int16_t)WEBRTC_SPL_SHIFT_W32(ener_comp, -shift3) *
+            crossSquareMax;
+
+        /* Calculate shift value, so that the two measures can
+           be put in the same Q domain */
+        if(2 * shiftMax + shift3 > 2 * shift1 + shift2) {
+          tmp1 =
+              WEBRTC_SPL_MIN(31, 2 * shiftMax + shift3 - 2 * shift1 - shift2);
+          tmp2 = 0;
+        } else {
+          tmp1 = 0;
+          tmp2 =
+              WEBRTC_SPL_MIN(31, 2 * shift1 + shift2 - 2 * shiftMax - shift3);
+        }
+
+        if ((measure>>tmp1) > (maxMeasure>>tmp2)) {
+          /* New lag is better => record lag, measure and domain */
+          lag = j;
+          crossSquareMax = crossSquare;
+          cross = cross_comp;
+          shiftMax = shift1;
+          ener = ener_comp;
+        }
+      }
+
+      /* Calculate the periodicity for the lag with the maximum correlation.
+
+         Definition of the periodicity:
+         abs(corr(vec1, vec2))/(sqrt(energy(vec1))*sqrt(energy(vec2)))
+
+         Work in the Square domain to simplify the calculations
+         max_perSquare is less than 1 (in Q15)
+      */
+      tmp2W32=WebRtcSpl_DotProductWithScale(&iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen],
+                                            &iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen],
+                                            corrLen, scale3);
+
+      if ((tmp2W32>0)&&(ener_comp>0)) {
+        /* norm energies to int16_t, compute the product of the energies and
+           use the upper int16_t as the denominator */
+
+        scale1=(int16_t)WebRtcSpl_NormW32(tmp2W32)-16;
+        tmp1=(int16_t)WEBRTC_SPL_SHIFT_W32(tmp2W32, scale1);
+
+        scale2=(int16_t)WebRtcSpl_NormW32(ener)-16;
+        tmp2=(int16_t)WEBRTC_SPL_SHIFT_W32(ener, scale2);
+        denom = (int16_t)((tmp1 * tmp2) >> 16);  /* in Q(scale1+scale2-16) */
+
+        /* Square the cross correlation and norm it such that max_perSquare
+           will be in Q15 after the division */
+
+        totscale = scale1+scale2-1;
+        tmp1 = (int16_t)WEBRTC_SPL_SHIFT_W32(cross, (totscale>>1));
+        tmp2 = (int16_t)WEBRTC_SPL_SHIFT_W32(cross, totscale-(totscale>>1));
+
+        nom = tmp1 * tmp2;
+        max_perSquare = (int16_t)WebRtcSpl_DivW32W16(nom, denom);
+
+      } else {
+        max_perSquare = 0;
+      }
+    }
+
+    /* previous frame lost, use recorded lag and gain */
+
+    else {
+      lag = iLBCdec_inst->prevLag;
+      max_perSquare = iLBCdec_inst->perSquare;
+    }
+
+    /* Attenuate signal and scale down pitch pred gain if
+       several frames lost consecutively */
+
+    use_gain = 32767;   /* 1.0 in Q15 */
+
+    if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>320) {
+      use_gain = 29491;  /* 0.9 in Q15 */
+    } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>640) {
+      use_gain = 22938;  /* 0.7 in Q15 */
+    } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>960) {
+      use_gain = 16384;  /* 0.5 in Q15 */
+    } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>1280) {
+      use_gain = 0;   /* 0.0 in Q15 */
+    }
+
+    /* Compute mixing factor of picth repeatition and noise:
+       for max_per>0.7 set periodicity to 1.0
+       0.4<max_per<0.7 set periodicity to (maxper-0.4)/0.7-0.4)
+       max_per<0.4 set periodicity to 0.0
+    */
+
+    if (max_perSquare>7868) { /* periodicity > 0.7  (0.7^4=0.2401 in Q15) */
+      pitchfact = 32767;
+    } else if (max_perSquare>839) { /* 0.4 < periodicity < 0.7 (0.4^4=0.0256 in Q15) */
+      /* find best index and interpolate from that */
+      ind = 5;
+      while ((max_perSquare<WebRtcIlbcfix_kPlcPerSqr[ind])&&(ind>0)) {
+        ind--;
+      }
+      /* pitch fact is approximated by first order */
+      tmpW32 = (int32_t)WebRtcIlbcfix_kPlcPitchFact[ind] +
+          ((WebRtcIlbcfix_kPlcPfSlope[ind] *
+              (max_perSquare - WebRtcIlbcfix_kPlcPerSqr[ind])) >> 11);
+
+      pitchfact = (int16_t)WEBRTC_SPL_MIN(tmpW32, 32767); /* guard against overflow */
+
+    } else { /* periodicity < 0.4 */
+      pitchfact = 0;
+    }
+
+    /* avoid repetition of same pitch cycle (buzzyness) */
+    use_lag = lag;
+    if (lag<80) {
+      use_lag = 2*lag;
+    }
+
+    /* compute concealed residual */
+    noise_energy_threshold_30dB = (int32_t)iLBCdec_inst->blockl * 900;
+    energy = 0;
+    for (i=0; i<iLBCdec_inst->blockl; i++) {
+
+      /* noise component -  52 < randlagFIX < 117 */
+      iLBCdec_inst->seed = (int16_t)(iLBCdec_inst->seed * 31821 + 13849);
+      randlag = 53 + (iLBCdec_inst->seed & 63);
+      if (randlag > i) {
+        randvec[i] =
+            iLBCdec_inst->prevResidual[iLBCdec_inst->blockl + i - randlag];
+      } else {
+        randvec[i] = iLBCdec_inst->prevResidual[i - randlag];
+      }
+
+      /* pitch repeatition component */
+      if (use_lag > i) {
+        PLCresidual[i] =
+            iLBCdec_inst->prevResidual[iLBCdec_inst->blockl + i - use_lag];
+      } else {
+        PLCresidual[i] = PLCresidual[i - use_lag];
+      }
+
+      /* Attinuate total gain for each 10 ms */
+      if (i<80) {
+        tot_gain=use_gain;
+      } else if (i<160) {
+        tot_gain = (int16_t)((31130 * use_gain) >> 15);  /* 0.95*use_gain */
+      } else {
+        tot_gain = (int16_t)((29491 * use_gain) >> 15);  /* 0.9*use_gain */
+      }
+
+
+      /* mix noise and pitch repeatition */
+      PLCresidual[i] = (int16_t)((tot_gain *
+          ((pitchfact * PLCresidual[i] + (32767 - pitchfact) * randvec[i] +
+              16384) >> 15)) >> 15);
+
+      /* Compute energy until threshold for noise energy is reached */
+      if (energy < noise_energy_threshold_30dB) {
+        energy += PLCresidual[i] * PLCresidual[i];
+      }
+    }
+
+    /* less than 30 dB, use only noise */
+    if (energy < noise_energy_threshold_30dB) {
+      for (i=0; i<iLBCdec_inst->blockl; i++) {
+        PLCresidual[i] = randvec[i];
+      }
+    }
+
+    /* use the old LPC */
+    WEBRTC_SPL_MEMCPY_W16(PLClpc, (*iLBCdec_inst).prevLpc, LPC_FILTERORDER+1);
+
+    /* Update state in case there are multiple frame losses */
+    iLBCdec_inst->prevLag = lag;
+    iLBCdec_inst->perSquare = max_perSquare;
+  }
+
+  /* no packet loss, copy input */
+
+  else {
+    WEBRTC_SPL_MEMCPY_W16(PLCresidual, decresidual, iLBCdec_inst->blockl);
+    WEBRTC_SPL_MEMCPY_W16(PLClpc, lpc, (LPC_FILTERORDER+1));
+    iLBCdec_inst->consPLICount = 0;
+  }
+
+  /* update state */
+  iLBCdec_inst->prevPLI = PLI;
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevLpc, PLClpc, (LPC_FILTERORDER+1));
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevResidual, PLCresidual, iLBCdec_inst->blockl);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.h
new file mode 100644
index 0000000000..c19c4eca32
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/do_plc.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_DoThePlc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Packet loss concealment routine. Conceals a residual signal
+ *  and LP parameters. If no packet loss, update state.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_DoThePlc(
+    int16_t* PLCresidual, /* (o) concealed residual */
+    int16_t* PLClpc,      /* (o) concealed LP parameters */
+    int16_t PLI,          /* (i) packet loss indicator
+                                                                0 - no PL, 1 = PL */
+    int16_t* decresidual, /* (i) decoded residual */
+    int16_t* lpc,         /* (i) decoded LPC (only used for no PL) */
+    size_t inlag,         /* (i) pitch lag */
+    IlbcDecoder* iLBCdec_inst
+    /* (i/o) decoder instance */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.c
new file mode 100644
index 0000000000..8e536221cd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.c
@@ -0,0 +1,517 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Encode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/encode.h"
+
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
+#include "modules/audio_coding/codecs/ilbc/cb_search.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/frame_classify.h"
+#include "modules/audio_coding/codecs/ilbc/hp_input.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_enc.h"
+#include "modules/audio_coding/codecs/ilbc/lpc_encode.h"
+#include "modules/audio_coding/codecs/ilbc/pack_bits.h"
+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
+#include "modules/audio_coding/codecs/ilbc/state_search.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/arch.h"
+
+#ifdef SPLIT_10MS
+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+#endif
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
+#endif
+
+/*----------------------------------------------------------------*
+ *  main encoder function
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EncodeImpl(
+    uint16_t *bytes,     /* (o) encoded data bits iLBC */
+    const int16_t *block, /* (i) speech vector to encode */
+    IlbcEncoder *iLBCenc_inst /* (i/o) the general encoder
+                                     state */
+                          ){
+  size_t n, meml_gotten, Nfor;
+  size_t diff, start_pos;
+  size_t index;
+  size_t subcount, subframe;
+  size_t start_count, end_count;
+  int16_t *residual;
+  int32_t en1, en2;
+  int16_t scale, max;
+  int16_t *syntdenum;
+  int16_t *decresidual;
+  int16_t *reverseResidual;
+  int16_t *reverseDecresidual;
+  /* Stack based */
+  int16_t weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
+  int16_t dataVec[BLOCKL_MAX + LPC_FILTERORDER];
+  int16_t memVec[CB_MEML+CB_FILTERLEN];
+  int16_t bitsMemory[sizeof(iLBC_bits)/sizeof(int16_t)];
+  iLBC_bits *iLBCbits_inst = (iLBC_bits*)bitsMemory;
+
+
+#ifdef SPLIT_10MS
+  int16_t *weightdenumbuf = iLBCenc_inst->weightdenumbuf;
+  int16_t last_bit;
+#endif
+
+  int16_t *data = &dataVec[LPC_FILTERORDER];
+  int16_t *mem = &memVec[CB_HALFFILTERLEN];
+
+  /* Reuse som buffers to save stack memory */
+  residual = &iLBCenc_inst->lpc_buffer[LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl];
+  syntdenum = mem;      /* syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX] and mem are used non overlapping in the code */
+  decresidual = residual;     /* Already encoded residual is overwritten by the decoded version */
+  reverseResidual = data;     /* data and reverseResidual are used non overlapping in the code */
+  reverseDecresidual = reverseResidual; /* Already encoded residual is overwritten by the decoded version */
+
+#ifdef SPLIT_10MS
+
+  WebRtcSpl_MemSetW16 (  (int16_t *) iLBCbits_inst, 0,
+                         sizeof(iLBC_bits) / sizeof(int16_t)  );
+
+  start_pos = iLBCenc_inst->start_pos;
+  diff = iLBCenc_inst->diff;
+
+  if (iLBCenc_inst->section != 0){
+    WEBRTC_SPL_MEMCPY_W16 (weightdenum, weightdenumbuf,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+    /* Un-Packetize the frame into parameters */
+    last_bit = WebRtcIlbcfix_UnpackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+    if (last_bit)
+      return;
+    /* adjust index */
+    WebRtcIlbcfix_IndexConvDec (iLBCbits_inst->cb_index);
+
+    if (iLBCenc_inst->section == 1){
+      /* Save first 80 samples of a 160/240 sample frame for 20/30msec */
+      WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples, block, 80);
+    }
+    else{ // iLBCenc_inst->section == 2 AND mode = 30ms
+      /* Save second 80 samples of a 240 sample frame for 30msec */
+      WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples + 80, block, 80);
+    }
+  }
+  else{ // iLBCenc_inst->section == 0
+    /* form a complete frame of 160/240 for 20msec/30msec mode */
+    WEBRTC_SPL_MEMCPY_W16 (data + (iLBCenc_inst->mode * 8) - 80, block, 80);
+    WEBRTC_SPL_MEMCPY_W16 (data, iLBCenc_inst->past_samples,
+                           (iLBCenc_inst->mode * 8) - 80);
+    iLBCenc_inst->Nfor_flag = 0;
+    iLBCenc_inst->Nback_flag = 0;
+#else
+    /* copy input block to data*/
+    WEBRTC_SPL_MEMCPY_W16(data,block,iLBCenc_inst->blockl);
+#endif
+
+    /* high pass filtering of input signal and scale down the residual (*0.5) */
+    WebRtcIlbcfix_HpInput(data, (int16_t*)WebRtcIlbcfix_kHpInCoefs,
+                          iLBCenc_inst->hpimemy, iLBCenc_inst->hpimemx,
+                          iLBCenc_inst->blockl);
+
+    /* LPC of hp filtered input data */
+    WebRtcIlbcfix_LpcEncode(syntdenum, weightdenum, iLBCbits_inst->lsf, data,
+                            iLBCenc_inst);
+
+    /* Set up state */
+    WEBRTC_SPL_MEMCPY_W16(dataVec, iLBCenc_inst->anaMem, LPC_FILTERORDER);
+
+    /* inverse filter to get residual */
+    for (n=0; n<iLBCenc_inst->nsub; n++ ) {
+      WebRtcSpl_FilterMAFastQ12(
+          &data[n*SUBL], &residual[n*SUBL],
+          &syntdenum[n*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, SUBL);
+    }
+
+    /* Copy the state for next frame */
+    WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->anaMem, &data[iLBCenc_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
+
+    /* find state location */
+
+    iLBCbits_inst->startIdx = WebRtcIlbcfix_FrameClassify(iLBCenc_inst,residual);
+
+    /* check if state should be in first or last part of the
+       two subframes */
+
+    index = (iLBCbits_inst->startIdx-1)*SUBL;
+    max=WebRtcSpl_MaxAbsValueW16(&residual[index], 2*SUBL);
+    scale = WebRtcSpl_GetSizeInBits((uint32_t)(max * max));
+
+    /* Scale to maximum 25 bits so that the MAC won't cause overflow */
+    scale = scale - 25;
+    if(scale < 0) {
+      scale = 0;
+    }
+
+    diff = STATE_LEN - iLBCenc_inst->state_short_len;
+    en1=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index],
+                                      iLBCenc_inst->state_short_len, scale);
+    index += diff;
+    en2=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index],
+                                      iLBCenc_inst->state_short_len, scale);
+    if (en1 > en2) {
+      iLBCbits_inst->state_first = 1;
+      start_pos = (iLBCbits_inst->startIdx-1)*SUBL;
+    } else {
+      iLBCbits_inst->state_first = 0;
+      start_pos = (iLBCbits_inst->startIdx-1)*SUBL + diff;
+    }
+
+    /* scalar quantization of state */
+
+    WebRtcIlbcfix_StateSearch(iLBCenc_inst, iLBCbits_inst, &residual[start_pos],
+                              &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+                              &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)]);
+
+    WebRtcIlbcfix_StateConstruct(iLBCbits_inst->idxForMax, iLBCbits_inst->idxVec,
+                                 &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+                                 &decresidual[start_pos], iLBCenc_inst->state_short_len
+                                 );
+
+    /* predictive quantization in state */
+
+    if (iLBCbits_inst->state_first) { /* put adaptive part in the end */
+
+      /* setup memory */
+
+      WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len);
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCenc_inst->state_short_len,
+                            decresidual+start_pos, iLBCenc_inst->state_short_len);
+
+      /* encode subframes */
+
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+                             &residual[start_pos+iLBCenc_inst->state_short_len],
+                             mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff,
+                             &weightdenum[iLBCbits_inst->startIdx*(LPC_FILTERORDER+1)], 0);
+
+      /* construct decoded vector */
+
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+          &decresidual[start_pos + iLBCenc_inst->state_short_len],
+          iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+          mem + CB_MEML - ST_MEM_L_TBL, ST_MEM_L_TBL, diff));
+
+    }
+    else { /* put adaptive part in the beginning */
+
+      /* create reversed vectors for prediction */
+
+      WebRtcSpl_MemCpyReversedOrder(&reverseResidual[diff-1],
+                                    &residual[(iLBCbits_inst->startIdx+1)*SUBL-STATE_LEN], diff);
+
+      /* setup memory */
+
+      meml_gotten = iLBCenc_inst->state_short_len;
+      WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[start_pos], meml_gotten);
+      WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len);
+
+      /* encode subframes */
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
+                             reverseResidual, mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff,
+                             &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
+                             0);
+
+      /* construct decoded vector */
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+            reverseDecresidual, iLBCbits_inst->cb_index,
+            iLBCbits_inst->gain_index, mem + CB_MEML - ST_MEM_L_TBL,
+            ST_MEM_L_TBL, diff));
+
+      /* get decoded residual from reversed vector */
+
+      WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1], reverseDecresidual, diff);
+    }
+
+#ifdef SPLIT_10MS
+    iLBCenc_inst->start_pos = start_pos;
+    iLBCenc_inst->diff = diff;
+    iLBCenc_inst->section++;
+    /* adjust index */
+    WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index);
+    /* Packetize the parameters into the frame */
+    WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+    WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+    return;
+  }
+#endif
+
+  /* forward prediction of subframes */
+
+  Nfor = iLBCenc_inst->nsub-iLBCbits_inst->startIdx-1;
+
+  /* counter for predicted subframes */
+#ifdef SPLIT_10MS
+  if (iLBCenc_inst->mode == 20)
+  {
+    subcount = 1;
+  }
+  if (iLBCenc_inst->mode == 30)
+  {
+    if (iLBCenc_inst->section == 1)
+    {
+      subcount = 1;
+    }
+    if (iLBCenc_inst->section == 2)
+    {
+      subcount = 3;
+    }
+  }
+#else
+  subcount=1;
+#endif
+
+  if( Nfor > 0 ){
+
+    /* setup memory */
+
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
+    WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN,
+                          decresidual+(iLBCbits_inst->startIdx-1)*SUBL, STATE_LEN);
+
+#ifdef SPLIT_10MS
+    if (iLBCenc_inst->Nfor_flag > 0)
+    {
+      for (subframe = 0; subframe < WEBRTC_SPL_MIN (Nfor, 2); subframe++)
+      {
+        /* update memory */
+        WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL));
+        WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL,
+                               &decresidual[(iLBCbits_inst->startIdx + 1 +
+                                             subframe) * SUBL], SUBL);
+      }
+    }
+
+    iLBCenc_inst->Nfor_flag++;
+
+    if (iLBCenc_inst->mode == 20)
+    {
+      start_count = 0;
+      end_count = Nfor;
+    }
+    if (iLBCenc_inst->mode == 30)
+    {
+      if (iLBCenc_inst->section == 1)
+      {
+        start_count = 0;
+        end_count = WEBRTC_SPL_MIN (Nfor, (size_t)2);
+      }
+      if (iLBCenc_inst->section == 2)
+      {
+        start_count = WEBRTC_SPL_MIN (Nfor, (size_t)2);
+        end_count = Nfor;
+      }
+    }
+#else
+    start_count = 0;
+    end_count = Nfor;
+#endif
+
+    /* loop over subframes to encode */
+
+    for (subframe = start_count; subframe < end_count; subframe++){
+
+      /* encode subframe */
+
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
+                             iLBCbits_inst->gain_index+subcount*CB_NSTAGES,
+                             &residual[(iLBCbits_inst->startIdx+1+subframe)*SUBL],
+                             mem, MEM_LF_TBL, SUBL,
+                             &weightdenum[(iLBCbits_inst->startIdx+1+subframe)*(LPC_FILTERORDER+1)],
+                             subcount);
+
+      /* construct decoded vector */
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+            &decresidual[(iLBCbits_inst->startIdx + 1 + subframe) * SUBL],
+            iLBCbits_inst->cb_index + subcount * CB_NSTAGES,
+            iLBCbits_inst->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+            SUBL));
+
+      /* update memory */
+
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &decresidual[(iLBCbits_inst->startIdx+1+subframe)*SUBL], SUBL);
+
+      subcount++;
+    }
+  }
+
+#ifdef SPLIT_10MS
+  if ((iLBCenc_inst->section == 1) &&
+      (iLBCenc_inst->mode == 30) && (Nfor > 0) && (end_count == 2))
+  {
+    iLBCenc_inst->section++;
+    /* adjust index */
+    WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index);
+    /* Packetize the parameters into the frame */
+    WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+    WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+    return;
+  }
+#endif
+
+  /* backward prediction of subframes */
+
+  if (iLBCbits_inst->startIdx > 1) {
+
+    /* create reverse order vectors
+       (The decresidual does not need to be copied since it is
+       contained in the same vector as the residual)
+    */
+
+    size_t Nback = iLBCbits_inst->startIdx - 1;
+    WebRtcSpl_MemCpyReversedOrder(&reverseResidual[Nback*SUBL-1], residual, Nback*SUBL);
+
+    /* setup memory */
+
+    meml_gotten = SUBL*(iLBCenc_inst->nsub+1-iLBCbits_inst->startIdx);
+    if( meml_gotten > CB_MEML ) {
+      meml_gotten=CB_MEML;
+    }
+
+    WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[Nback*SUBL], meml_gotten);
+    WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
+
+#ifdef SPLIT_10MS
+    if (iLBCenc_inst->Nback_flag > 0)
+    {
+      for (subframe = 0; subframe < WEBRTC_SPL_MAX (2 - Nfor, 0); subframe++)
+      {
+        /* update memory */
+        WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL));
+        WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL,
+                               &reverseDecresidual[subframe * SUBL], SUBL);
+      }
+    }
+
+    iLBCenc_inst->Nback_flag++;
+
+
+    if (iLBCenc_inst->mode == 20)
+    {
+      start_count = 0;
+      end_count = Nback;
+    }
+    if (iLBCenc_inst->mode == 30)
+    {
+      if (iLBCenc_inst->section == 1)
+      {
+        start_count = 0;
+        end_count = (Nfor >= 2) ? 0 : (2 - NFor);
+      }
+      if (iLBCenc_inst->section == 2)
+      {
+        start_count = (Nfor >= 2) ? 0 : (2 - NFor);
+        end_count = Nback;
+      }
+    }
+#else
+    start_count = 0;
+    end_count = Nback;
+#endif
+
+    /* loop over subframes to encode */
+
+    for (subframe = start_count; subframe < end_count; subframe++){
+
+      /* encode subframe */
+
+      WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
+                             iLBCbits_inst->gain_index+subcount*CB_NSTAGES, &reverseResidual[subframe*SUBL],
+                             mem, MEM_LF_TBL, SUBL,
+                             &weightdenum[(iLBCbits_inst->startIdx-2-subframe)*(LPC_FILTERORDER+1)],
+                             subcount);
+
+      /* construct decoded vector */
+      RTC_CHECK(WebRtcIlbcfix_CbConstruct(
+            &reverseDecresidual[subframe * SUBL],
+            iLBCbits_inst->cb_index + subcount * CB_NSTAGES,
+            iLBCbits_inst->gain_index + subcount * CB_NSTAGES, mem, MEM_LF_TBL,
+            SUBL));
+
+      /* update memory */
+      memmove(mem, mem + SUBL, (CB_MEML - SUBL) * sizeof(*mem));
+      WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
+                            &reverseDecresidual[subframe*SUBL], SUBL);
+
+      subcount++;
+
+    }
+
+    /* get decoded residual from reversed vector */
+
+    WebRtcSpl_MemCpyReversedOrder(&decresidual[SUBL*Nback-1], reverseDecresidual, SUBL*Nback);
+  }
+  /* end encoding part */
+
+  /* adjust index */
+
+  WebRtcIlbcfix_IndexConvEnc(iLBCbits_inst->cb_index);
+
+  /* Packetize the parameters into the frame */
+
+#ifdef SPLIT_10MS
+  if( (iLBCenc_inst->mode==30) && (iLBCenc_inst->section==1) ){
+    WebRtcIlbcfix_PackBits(iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
+  }
+  else{
+    WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
+  }
+#else
+  WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
+#endif
+
+#ifndef WEBRTC_ARCH_BIG_ENDIAN
+  /* Swap bytes for LITTLE ENDIAN since the packbits()
+     function assumes BIG_ENDIAN machine */
+#ifdef SPLIT_10MS
+  if (( (iLBCenc_inst->section == 1) && (iLBCenc_inst->mode == 20) ) ||
+      ( (iLBCenc_inst->section == 2) && (iLBCenc_inst->mode == 30) )){
+    WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words, bytes);
+  }
+#else
+  WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words, bytes);
+#endif
+#endif
+
+#ifdef SPLIT_10MS
+  if (subcount == (iLBCenc_inst->nsub - 1))
+  {
+    iLBCenc_inst->section = 0;
+  }
+  else
+  {
+    iLBCenc_inst->section++;
+    WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
+                           SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
+  }
+#endif
+
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.h
new file mode 100644
index 0000000000..bc3e187d92
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/encode.h
@@ -0,0 +1,38 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Encode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  main encoder function
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EncodeImpl(
+    uint16_t* bytes,      /* (o) encoded data bits iLBC */
+    const int16_t* block, /* (i) speech vector to encode */
+    IlbcEncoder* iLBCenc_inst /* (i/o) the general encoder
+                                           state */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
new file mode 100644
index 0000000000..7f00254aea
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnergyInverse.c
+
+******************************************************************/
+
+/* Inverses the in vector in into Q29 domain */
+
+#include "modules/audio_coding/codecs/ilbc/energy_inverse.h"
+
+void WebRtcIlbcfix_EnergyInverse(
+    int16_t *energy,    /* (i/o) Energy and inverse
+                                                           energy (in Q29) */
+    size_t noOfEnergies)  /* (i)   The length of the energy
+                                   vector */
+{
+  int32_t Nom=(int32_t)0x1FFFFFFF;
+  int16_t *energyPtr;
+  size_t i;
+
+  /* Set the minimum energy value to 16384 to avoid overflow */
+  energyPtr=energy;
+  for (i=0; i<noOfEnergies; i++) {
+    (*energyPtr)=WEBRTC_SPL_MAX((*energyPtr),16384);
+    energyPtr++;
+  }
+
+  /* Calculate inverse energy in Q29 */
+  energyPtr=energy;
+  for (i=0; i<noOfEnergies; i++) {
+    (*energyPtr) = (int16_t)WebRtcSpl_DivW32W16(Nom, (*energyPtr));
+    energyPtr++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
new file mode 100644
index 0000000000..15391cf230
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnergyInverse.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/* Inverses the in vector in into Q29 domain */
+
+void WebRtcIlbcfix_EnergyInverse(
+    int16_t*
+        energy, /* (i/o) Energy and inverse
+                                                          energy (in Q29) */
+    size_t noOfEnergies); /* (i)   The length of the energy
+                                 vector */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c
new file mode 100644
index 0000000000..cd3d0a4db1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhUpsample.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/enh_upsample.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * upsample finite array assuming zeros outside bounds
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EnhUpsample(
+    int32_t *useq1, /* (o) upsampled output sequence */
+    int16_t *seq1 /* (i) unupsampled sequence */
+                                ){
+  int j;
+  int32_t *pu1, *pu11;
+  int16_t *ps, *w16tmp;
+  const int16_t *pp;
+
+  /* filtering: filter overhangs left side of sequence */
+  pu1=useq1;
+  for (j=0;j<ENH_UPS0; j++) {
+    pu11=pu1;
+    /* i = 2 */
+    pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+    ps=seq1+2;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+    /* i = 3 */
+    pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+    ps=seq1+3;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+    /* i = 4 */
+    pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
+    ps=seq1+4;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu1++;
+  }
+
+  /* filtering: simple convolution=inner products
+     (not needed since the sequence is so short)
+  */
+
+  /* filtering: filter overhangs right side of sequence */
+
+  /* Code with loops, which is equivivalent to the expanded version below
+
+     filterlength = 5;
+     hf1 = 2;
+     for(j=0;j<ENH_UPS0; j++){
+     pu = useq1 + (filterlength-hfl)*ENH_UPS0 + j;
+     for(i=1; i<=hfl; i++){
+     *pu=0;
+     pp = polyp[j]+i;
+     ps = seq1+dim1-1;
+     for(k=0;k<filterlength-i;k++) {
+     *pu += (*ps--) * *pp++;
+     }
+     pu+=ENH_UPS0;
+     }
+     }
+  */
+  pu1 = useq1 + 12;
+  w16tmp = seq1+4;
+  for (j=0;j<ENH_UPS0; j++) {
+    pu11 = pu1;
+    /* i = 1 */
+    pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+2;
+    ps = w16tmp;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+    /* i = 2 */
+    pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+3;
+    ps = w16tmp;
+    *pu11 = (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    *pu11 += (*ps--) * *pp++;
+    pu11+=ENH_UPS0;
+
+    pu1++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h
new file mode 100644
index 0000000000..b427eca50a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhUpsample.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * upsample finite array assuming zeros outside bounds
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_EnhUpsample(
+    int32_t* useq1, /* (o) upsampled output sequence */
+    int16_t* seq1   /* (i) unupsampled sequence */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.c
new file mode 100644
index 0000000000..bd4e60015c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.c
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Enhancer.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/enhancer.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/get_sync_seq.h"
+#include "modules/audio_coding/codecs/ilbc/smooth.h"
+
+/*----------------------------------------------------------------*
+ * perform enhancement on idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Enhancer(
+    int16_t *odata,   /* (o) smoothed block, dimension blockl */
+    int16_t *idata,   /* (i) data buffer used for enhancing */
+    size_t idatal,   /* (i) dimension idata */
+    size_t centerStartPos, /* (i) first sample current block within idata */
+    size_t *period,   /* (i) pitch period array (pitch bward-in time) */
+    const size_t *plocs,   /* (i) locations where period array values valid */
+    size_t periodl   /* (i) dimension of period and plocs */
+                            ){
+  /* Stack based */
+  int16_t surround[ENH_BLOCKL];
+
+  WebRtcSpl_MemSetW16(surround, 0, ENH_BLOCKL);
+
+  /* get said second sequence of segments */
+
+  WebRtcIlbcfix_GetSyncSeq(idata, idatal, centerStartPos, period, plocs,
+                           periodl, ENH_HL, surround);
+
+  /* compute the smoothed output from said second sequence */
+
+  WebRtcIlbcfix_Smooth(odata, idata + centerStartPos, surround);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.h
new file mode 100644
index 0000000000..386949347a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Enhancer.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * perform enhancement on idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Enhancer(
+    int16_t* odata,        /* (o) smoothed block, dimension blockl */
+    int16_t* idata,        /* (i) data buffer used for enhancing */
+    size_t idatal,         /* (i) dimension idata */
+    size_t centerStartPos, /* (i) first sample current block within idata */
+    size_t* period,        /* (i) pitch period array (pitch bward-in time) */
+    const size_t* plocs,   /* (i) locations where period array values valid */
+    size_t periodl         /* (i) dimension of period and plocs */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
new file mode 100644
index 0000000000..ca23e19ae3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -0,0 +1,382 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhancerInterface.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/enhancer.h"
+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+
+
+
+/*----------------------------------------------------------------*
+ * interface for enhancer
+ *---------------------------------------------------------------*/
+
+size_t  // (o) Estimated lag in end of in[]
+    WebRtcIlbcfix_EnhancerInterface(
+        int16_t* out,                 // (o) enhanced signal
+        const int16_t* in,            // (i) unenhanced signal
+        IlbcDecoder* iLBCdec_inst) {  // (i) buffers etc
+  size_t iblock;
+  size_t lag=20, tlag=20;
+  size_t inLen=iLBCdec_inst->blockl+120;
+  int16_t scale, scale1;
+  size_t plc_blockl;
+  int16_t *enh_buf;
+  size_t *enh_period;
+  int32_t tmp1, tmp2, max;
+  size_t new_blocks;
+  int16_t *enh_bufPtr1;
+  size_t i;
+  size_t k;
+  int16_t EnChange;
+  int16_t SqrtEnChange;
+  int16_t inc;
+  int16_t win;
+  int16_t *tmpW16ptr;
+  size_t startPos;
+  int16_t *plc_pred;
+  const int16_t *target, *regressor;
+  int16_t max16;
+  int shifts;
+  int32_t ener;
+  int16_t enerSh;
+  int16_t corrSh;
+  size_t ind;
+  int16_t sh;
+  size_t start, stop;
+  /* Stack based */
+  int16_t totsh[3];
+  int16_t downsampled[(BLOCKL_MAX+120)>>1]; /* length 180 */
+  int32_t corr32[50];
+  int32_t corrmax[3];
+  int16_t corr16[3];
+  int16_t en16[3];
+  size_t lagmax[3];
+
+  plc_pred = downsampled; /* Reuse memory since plc_pred[ENH_BLOCKL] and
+                              downsampled are non overlapping */
+  enh_buf=iLBCdec_inst->enh_buf;
+  enh_period=iLBCdec_inst->enh_period;
+
+  /* Copy in the new data into the enhancer buffer */
+  memmove(enh_buf, &enh_buf[iLBCdec_inst->blockl],
+          (ENH_BUFL - iLBCdec_inst->blockl) * sizeof(*enh_buf));
+
+  WEBRTC_SPL_MEMCPY_W16(&enh_buf[ENH_BUFL-iLBCdec_inst->blockl], in,
+                        iLBCdec_inst->blockl);
+
+  /* Set variables that are dependent on frame size */
+  if (iLBCdec_inst->mode==30) {
+    plc_blockl=ENH_BLOCKL;
+    new_blocks=3;
+    startPos=320;  /* Start position for enhancement
+                     (640-new_blocks*ENH_BLOCKL-80) */
+  } else {
+    plc_blockl=40;
+    new_blocks=2;
+    startPos=440;  /* Start position for enhancement
+                    (640-new_blocks*ENH_BLOCKL-40) */
+  }
+
+  /* Update the pitch prediction for each enhancer block, move the old ones */
+  memmove(enh_period, &enh_period[new_blocks],
+          (ENH_NBLOCKS_TOT - new_blocks) * sizeof(*enh_period));
+
+  WebRtcSpl_DownsampleFast(
+      enh_buf+ENH_BUFL-inLen,    /* Input samples */
+      inLen + ENH_BUFL_FILTEROVERHEAD,
+      downsampled,
+      inLen / 2,
+      (int16_t*)WebRtcIlbcfix_kLpFiltCoefs,  /* Coefficients in Q12 */
+      FILTERORDER_DS_PLUS1,    /* Length of filter (order-1) */
+      FACTOR_DS,
+      DELAY_DS);
+
+  /* Estimate the pitch in the down sampled domain. */
+  for(iblock = 0; iblock<new_blocks; iblock++){
+
+    /* references */
+    target = downsampled + 60 + iblock * ENH_BLOCKL_HALF;
+    regressor = target - 10;
+
+    /* scaling */
+    max16 = WebRtcSpl_MaxAbsValueW16(&regressor[-50], ENH_BLOCKL_HALF + 50 - 1);
+    shifts = WebRtcSpl_GetSizeInBits((uint32_t)(max16 * max16)) - 25;
+    shifts = WEBRTC_SPL_MAX(0, shifts);
+
+    /* compute cross correlation */
+    WebRtcSpl_CrossCorrelation(corr32, target, regressor, ENH_BLOCKL_HALF, 50,
+                               shifts, -1);
+
+    /* Find 3 highest correlations that should be compared for the
+       highest (corr*corr)/ener */
+
+    for (i=0;i<2;i++) {
+      lagmax[i] = WebRtcSpl_MaxIndexW32(corr32, 50);
+      corrmax[i] = corr32[lagmax[i]];
+      start = WEBRTC_SPL_MAX(2, lagmax[i]) - 2;
+      stop = WEBRTC_SPL_MIN(47, lagmax[i]) + 2;
+      for (k = start; k <= stop; k++) {
+        corr32[k] = 0;
+      }
+    }
+    lagmax[2] = WebRtcSpl_MaxIndexW32(corr32, 50);
+    corrmax[2] = corr32[lagmax[2]];
+
+    /* Calculate normalized corr^2 and ener */
+    for (i=0;i<3;i++) {
+      corrSh = 15-WebRtcSpl_GetSizeInBits(corrmax[i]);
+      ener = WebRtcSpl_DotProductWithScale(regressor - lagmax[i],
+                                           regressor - lagmax[i],
+                                           ENH_BLOCKL_HALF, shifts);
+      enerSh = 15-WebRtcSpl_GetSizeInBits(ener);
+      corr16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(corrmax[i], corrSh);
+      corr16[i] = (int16_t)((corr16[i] * corr16[i]) >> 16);
+      en16[i] = (int16_t)WEBRTC_SPL_SHIFT_W32(ener, enerSh);
+      totsh[i] = enerSh - 2 * corrSh;
+    }
+
+    /* Compare lagmax[0..3] for the (corr^2)/ener criteria */
+    ind = 0;
+    for (i=1; i<3; i++) {
+      if (totsh[ind] > totsh[i]) {
+        sh = WEBRTC_SPL_MIN(31, totsh[ind]-totsh[i]);
+        if (corr16[ind] * en16[i] < (corr16[i] * en16[ind]) >> sh) {
+          ind = i;
+        }
+      } else {
+        sh = WEBRTC_SPL_MIN(31, totsh[i]-totsh[ind]);
+        if ((corr16[ind] * en16[i]) >> sh < corr16[i] * en16[ind]) {
+          ind = i;
+        }
+      }
+    }
+
+    lag = lagmax[ind] + 10;
+
+    /* Store the estimated lag in the non-downsampled domain */
+    enh_period[ENH_NBLOCKS_TOT - new_blocks + iblock] = lag * 8;
+
+    /* Store the estimated lag for backward PLC */
+    if (iLBCdec_inst->prev_enh_pl==1) {
+      if (!iblock) {
+        tlag = lag * 2;
+      }
+    } else {
+      if (iblock==1) {
+        tlag = lag * 2;
+      }
+    }
+
+    lag *= 2;
+  }
+
+  if ((iLBCdec_inst->prev_enh_pl==1)||(iLBCdec_inst->prev_enh_pl==2)) {
+
+    /* Calculate the best lag of the new frame
+       This is used to interpolate backwards and mix with the PLC'd data
+    */
+
+    /* references */
+    target=in;
+    regressor=in+tlag-1;
+
+    /* scaling */
+    // Note that this is not abs-max, so we will take the absolute value below.
+    max16 = WebRtcSpl_MaxAbsElementW16(regressor, plc_blockl + 3 - 1);
+    const int16_t max_target =
+        WebRtcSpl_MaxAbsElementW16(target, plc_blockl + 3 - 1);
+    const int64_t max_val = plc_blockl * abs(max16 * max_target);
+    const int32_t factor = max_val >> 31;
+    shifts = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+
+    /* compute cross correlation */
+    WebRtcSpl_CrossCorrelation(corr32, target, regressor, plc_blockl, 3, shifts,
+                               1);
+
+    /* find lag */
+    lag=WebRtcSpl_MaxIndexW32(corr32, 3);
+    lag+=tlag-1;
+
+    /* Copy the backward PLC to plc_pred */
+
+    if (iLBCdec_inst->prev_enh_pl==1) {
+      if (lag>plc_blockl) {
+        WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-plc_blockl], plc_blockl);
+      } else {
+        WEBRTC_SPL_MEMCPY_W16(&plc_pred[plc_blockl-lag], in, lag);
+        WEBRTC_SPL_MEMCPY_W16(
+            plc_pred, &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl+lag],
+            (plc_blockl-lag));
+      }
+    } else {
+      size_t pos;
+
+      pos = plc_blockl;
+
+      while (lag<pos) {
+        WEBRTC_SPL_MEMCPY_W16(&plc_pred[pos-lag], in, lag);
+        pos = pos - lag;
+      }
+      WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-pos], pos);
+
+    }
+
+    if (iLBCdec_inst->prev_enh_pl==1) {
+      /* limit energy change
+         if energy in backward PLC is more than 4 times higher than the forward
+         PLC, then reduce the energy in the backward PLC vector:
+         sample 1...len-16 set energy of the to 4 times forward PLC
+         sample len-15..len interpolate between 4 times fw PLC and bw PLC energy
+
+         Note: Compared to floating point code there is a slight change,
+         the window is 16 samples long instead of 10 samples to simplify the
+         calculations
+      */
+
+      max=WebRtcSpl_MaxAbsValueW16(
+          &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl], plc_blockl);
+      max16=WebRtcSpl_MaxAbsValueW16(plc_pred, plc_blockl);
+      max = WEBRTC_SPL_MAX(max, max16);
+      scale=22-(int16_t)WebRtcSpl_NormW32(max);
+      scale=WEBRTC_SPL_MAX(scale,0);
+
+      tmp2 = WebRtcSpl_DotProductWithScale(
+          &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl],
+          &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl],
+          plc_blockl, scale);
+      tmp1 = WebRtcSpl_DotProductWithScale(plc_pred, plc_pred,
+                                           plc_blockl, scale);
+
+      /* Check the energy difference */
+      if ((tmp1>0)&&((tmp1>>2)>tmp2)) {
+        /* EnChange is now guaranteed to be <0.5
+           Calculate EnChange=tmp2/tmp1 in Q16
+        */
+
+        scale1=(int16_t)WebRtcSpl_NormW32(tmp1);
+        tmp1=WEBRTC_SPL_SHIFT_W32(tmp1, (scale1-16)); /* using 15 bits */
+
+        tmp2=WEBRTC_SPL_SHIFT_W32(tmp2, (scale1));
+        EnChange = (int16_t)WebRtcSpl_DivW32W16(tmp2,
+                                                      (int16_t)tmp1);
+
+        /* Calculate the Sqrt of the energy in Q15 ((14+16)/2) */
+        SqrtEnChange = (int16_t)WebRtcSpl_SqrtFloor(EnChange << 14);
+
+
+        /* Multiply first part of vector with 2*SqrtEnChange */
+        WebRtcSpl_ScaleVector(plc_pred, plc_pred, SqrtEnChange, plc_blockl-16,
+                              14);
+
+        /* Calculate increase parameter for window part (16 last samples) */
+        /* (1-2*SqrtEnChange)/16 in Q15 */
+        inc = 2048 - (SqrtEnChange >> 3);
+
+        win=0;
+        tmpW16ptr=&plc_pred[plc_blockl-16];
+
+        for (i=16;i>0;i--) {
+          *tmpW16ptr = (int16_t)(
+              (*tmpW16ptr * (SqrtEnChange + (win >> 1))) >> 14);
+          /* multiply by (2.0*SqrtEnChange+win) */
+
+          win += inc;
+          tmpW16ptr++;
+        }
+      }
+
+      /* Make the linear interpolation between the forward PLC'd data
+         and the backward PLC'd data (from the new frame)
+      */
+
+      if (plc_blockl==40) {
+        inc=400; /* 1/41 in Q14 */
+      } else { /* plc_blockl==80 */
+        inc=202; /* 1/81 in Q14 */
+      }
+      win=0;
+      enh_bufPtr1=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
+      for (i=0; i<plc_blockl; i++) {
+        win+=inc;
+        *enh_bufPtr1 = (int16_t)((*enh_bufPtr1 * win) >> 14);
+        *enh_bufPtr1 += (int16_t)(
+            ((16384 - win) * plc_pred[plc_blockl - 1 - i]) >> 14);
+        enh_bufPtr1--;
+      }
+    } else {
+      int16_t *synt = &downsampled[LPC_FILTERORDER];
+
+      enh_bufPtr1=&enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl];
+      WEBRTC_SPL_MEMCPY_W16(enh_bufPtr1, plc_pred, plc_blockl);
+
+      /* Clear fileter memory */
+      WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER);
+      WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4);
+      WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2);
+
+      /* Initialize filter memory by filtering through 2 lags */
+      WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], iLBCdec_inst->syntMem,
+                            LPC_FILTERORDER);
+      WebRtcSpl_FilterARFastQ12(
+          enh_bufPtr1,
+          synt,
+          &iLBCdec_inst->old_syntdenum[
+              (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, lag);
+
+      WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], &synt[lag-LPC_FILTERORDER],
+                            LPC_FILTERORDER);
+      WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
+                             iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+                             lag);
+      WebRtcSpl_FilterARFastQ12(
+          enh_bufPtr1, synt,
+          &iLBCdec_inst->old_syntdenum[
+              (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
+          LPC_FILTERORDER+1, lag);
+
+      WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &synt[lag-LPC_FILTERORDER],
+                            LPC_FILTERORDER);
+      WebRtcIlbcfix_HpOutput(synt, (int16_t*)WebRtcIlbcfix_kHpOutCoefs,
+                             iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
+                             lag);
+    }
+  }
+
+
+  /* Perform enhancement block by block */
+
+  for (iblock = 0; iblock<new_blocks; iblock++) {
+    WebRtcIlbcfix_Enhancer(out + iblock * ENH_BLOCKL,
+                           enh_buf,
+                           ENH_BUFL,
+                           iblock * ENH_BLOCKL + startPos,
+                           enh_period,
+                           WebRtcIlbcfix_kEnhPlocs, ENH_NBLOCKS_TOT);
+  }
+
+  return (lag);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
new file mode 100644
index 0000000000..5022a47c3a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_EnhancerInterface.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * interface for enhancer
+ *---------------------------------------------------------------*/
+
+size_t  // (o) Estimated lag in end of in[]
+WebRtcIlbcfix_EnhancerInterface(int16_t* out,       // (o) enhanced signal
+                                const int16_t* in,  // (i) unenhanced signal
+                                IlbcDecoder* iLBCdec_inst);  // (i) buffers etc
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
new file mode 100644
index 0000000000..6b4f30c96b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FilteredCbVecs.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Construct an additional codebook vector by filtering the
+ *  initial codebook buffer. This vector is then used to expand
+ *  the codebook with an additional section.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_FilteredCbVecs(
+    int16_t *cbvectors, /* (o) Codebook vector for the higher section */
+    int16_t *CBmem,  /* (i) Codebook memory that is filtered to create a
+                                           second CB section */
+    size_t lMem,  /* (i) Length of codebook memory */
+    size_t samples    /* (i) Number of samples to filter */
+                                  ) {
+
+  /* Set up the memory, start with zero state */
+  WebRtcSpl_MemSetW16(CBmem+lMem, 0, CB_HALFFILTERLEN);
+  WebRtcSpl_MemSetW16(CBmem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN);
+  WebRtcSpl_MemSetW16(cbvectors, 0, lMem-samples);
+
+  /* Filter to obtain the filtered CB memory */
+
+  WebRtcSpl_FilterMAFastQ12(
+      CBmem+CB_HALFFILTERLEN+lMem-samples, cbvectors+lMem-samples,
+      (int16_t*)WebRtcIlbcfix_kCbFiltersRev, CB_FILTERLEN, samples);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
new file mode 100644
index 0000000000..661262e42e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FilteredCbVecs.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Construct an additional codebook vector by filtering the
+ *  initial codebook buffer. This vector is then used to expand
+ *  the codebook with an additional section.
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_FilteredCbVecs(
+    int16_t* cbvectors, /* (o) Codebook vector for the higher section */
+    int16_t* CBmem,     /* (i) Codebook memory that is filtered to create a
+                                              second CB section */
+    size_t lMem,        /* (i) Length of codebook memory */
+    size_t samples      /* (i) Number of samples to filter */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
new file mode 100644
index 0000000000..c1084b1645
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FrameClassify.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/frame_classify.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Classification of subframes to localize start state
+ *---------------------------------------------------------------*/
+
+size_t WebRtcIlbcfix_FrameClassify(
+    /* (o) Index to the max-energy sub frame */
+    IlbcEncoder *iLBCenc_inst,
+    /* (i/o) the encoder state structure */
+    int16_t *residualFIX /* (i) lpc residual signal */
+                                                ){
+  int16_t max, scale;
+  int32_t ssqEn[NSUB_MAX-1];
+  int16_t *ssqPtr;
+  int32_t *seqEnPtr;
+  int32_t maxW32;
+  int16_t scale1;
+  size_t pos;
+  size_t n;
+
+  /*
+    Calculate the energy of each of the 80 sample blocks
+    in the draft the 4 first and last samples are windowed with 1/5...4/5
+    and 4/5...1/5 respectively. To simplify for the fixpoint we have changed
+    this to 0 0 1 1 and 1 1 0 0
+  */
+
+  max = WebRtcSpl_MaxAbsValueW16(residualFIX, iLBCenc_inst->blockl);
+  scale = WebRtcSpl_GetSizeInBits((uint32_t)(max * max));
+
+  /* Scale to maximum 24 bits so that it won't overflow for 76 samples */
+  scale = scale-24;
+  scale1 = WEBRTC_SPL_MAX(0, scale);
+
+  /* Calculate energies */
+  ssqPtr=residualFIX + 2;
+  seqEnPtr=ssqEn;
+  for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
+    (*seqEnPtr) = WebRtcSpl_DotProductWithScale(ssqPtr, ssqPtr, 76, scale1);
+    ssqPtr += 40;
+    seqEnPtr++;
+  }
+
+  /* Scale to maximum 20 bits in order to allow for the 11 bit window */
+  maxW32 = WebRtcSpl_MaxValueW32(ssqEn, iLBCenc_inst->nsub - 1);
+  scale = WebRtcSpl_GetSizeInBits(maxW32) - 20;
+  scale1 = WEBRTC_SPL_MAX(0, scale);
+
+  /* Window each 80 block with the ssqEn_winTbl window to give higher probability for
+     the blocks in the middle
+  */
+  seqEnPtr=ssqEn;
+  if (iLBCenc_inst->mode==20) {
+    ssqPtr=(int16_t*)WebRtcIlbcfix_kStartSequenceEnrgWin+1;
+  } else {
+    ssqPtr=(int16_t*)WebRtcIlbcfix_kStartSequenceEnrgWin;
+  }
+  for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
+    (*seqEnPtr)=WEBRTC_SPL_MUL(((*seqEnPtr)>>scale1), (*ssqPtr));
+    seqEnPtr++;
+    ssqPtr++;
+  }
+
+  /* Extract the best choise of start state */
+  pos = WebRtcSpl_MaxIndexW32(ssqEn, iLBCenc_inst->nsub - 1) + 1;
+
+  return(pos);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
new file mode 100644
index 0000000000..7615106d70
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_FrameClassify.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+size_t WebRtcIlbcfix_FrameClassify(
+    /* (o) Index to the max-energy sub frame */
+    IlbcEncoder* iLBCenc_inst,
+    /* (i/o) the encoder state structure */
+    int16_t* residualFIX /* (i) lpc residual signal */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c
new file mode 100644
index 0000000000..1357dece33
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainDequant.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/gain_dequant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  decoder for quantized gains in the gain-shape coding of
+ *  residual
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_GainDequant(
+    /* (o) quantized gain value (Q14) */
+    int16_t index, /* (i) quantization index */
+    int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */
+    int16_t stage /* (i) The stage of the search */
+                                                ){
+  int16_t scale;
+  const int16_t *gain;
+
+  /* obtain correct scale factor */
+
+  scale=WEBRTC_SPL_ABS_W16(maxIn);
+  scale = WEBRTC_SPL_MAX(1638, scale);  /* if lower than 0.1, set it to 0.1 */
+
+  /* select the quantization table and return the decoded value */
+  gain = WebRtcIlbcfix_kGain[stage];
+
+  return (int16_t)((scale * gain[index] + 8192) >> 14);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h
new file mode 100644
index 0000000000..2b97550b6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainDequant.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  decoder for quantized gains in the gain-shape coding of
+ *  residual
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_GainDequant(
+    /* (o) quantized gain value (Q14) */
+    int16_t index, /* (i) quantization index */
+    int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */
+    int16_t stage  /* (i) The stage of the search */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.c
new file mode 100644
index 0000000000..9a6d49d51a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.c
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainQuant.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/gain_quant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  quantizer for the gain in the gain-shape coding of residual
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
+    int16_t gain, /* (i) gain value Q14 */
+    int16_t maxIn, /* (i) maximum of gain value Q14 */
+    int16_t stage, /* (i) The stage of the search */
+    int16_t *index /* (o) quantization index */
+                                        ) {
+
+  int16_t scale, cblen;
+  int32_t gainW32, measure1, measure2;
+  const int16_t *cbPtr, *cb;
+  int loc, noMoves, noChecks, i;
+
+  /* ensure a lower bound (0.1) on the scaling factor */
+
+  scale = WEBRTC_SPL_MAX(1638, maxIn);
+
+  /* select the quantization table and calculate
+     the length of the table and the number of
+     steps in the binary search that are needed */
+  cb = WebRtcIlbcfix_kGain[stage];
+  cblen = 32>>stage;
+  noChecks = 4-stage;
+
+  /* Multiply the gain with 2^14 to make the comparison
+     easier and with higher precision */
+  gainW32 = gain << 14;
+
+  /* Do a binary search, starting in the middle of the CB
+     loc - defines the current position in the table
+     noMoves - defines the number of steps to move in the CB in order
+     to get next CB location
+  */
+
+  loc = cblen>>1;
+  noMoves = loc;
+  cbPtr = cb + loc; /* Centre of CB */
+
+  for (i=noChecks;i>0;i--) {
+    noMoves>>=1;
+    measure1 = scale * *cbPtr;
+
+    /* Move up if gain is larger, otherwise move down in table */
+    measure1 = measure1 - gainW32;
+
+    if (0>measure1) {
+      cbPtr+=noMoves;
+      loc+=noMoves;
+    } else {
+      cbPtr-=noMoves;
+      loc-=noMoves;
+    }
+  }
+
+  /* Check which value is the closest one: loc-1, loc or loc+1 */
+
+  measure1 = scale * *cbPtr;
+  if (gainW32>measure1) {
+    /* Check against value above loc */
+    measure2 = scale * cbPtr[1];
+    if ((measure2-gainW32)<(gainW32-measure1)) {
+      loc+=1;
+    }
+  } else {
+    /* Check against value below loc */
+    measure2 = scale * cbPtr[-1];
+    if ((gainW32-measure2)<=(measure1-gainW32)) {
+      loc-=1;
+    }
+  }
+
+  /* Guard against getting outside the table. The calculation above can give a location
+     which is one above the maximum value (in very rare cases) */
+  loc=WEBRTC_SPL_MIN(loc, (cblen-1));
+  *index=loc;
+
+  /* Calculate and return the quantized gain value (in Q14) */
+  return (int16_t)((scale * cb[loc] + 8192) >> 14);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.h
new file mode 100644
index 0000000000..761f7d2f79
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/gain_quant.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GainQuant.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  quantizer for the gain in the gain-shape coding of residual
+ *---------------------------------------------------------------*/
+
+int16_t
+WebRtcIlbcfix_GainQuant(               /* (o) quantized gain value */
+                        int16_t gain,  /* (i) gain value Q14 */
+                        int16_t maxIn, /* (i) maximum of gain value Q14 */
+                        int16_t stage, /* (i) The stage of the search */
+                        int16_t* index /* (o) quantization index */
+                        );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
new file mode 100644
index 0000000000..e9cd2008e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetCbVec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/get_cd_vec.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Construct codebook vector for given index.
+ *---------------------------------------------------------------*/
+
+bool WebRtcIlbcfix_GetCbVec(
+    int16_t *cbvec,   /* (o) Constructed codebook vector */
+    int16_t *mem,   /* (i) Codebook buffer */
+    size_t index,   /* (i) Codebook index */
+    size_t lMem,   /* (i) Length of codebook buffer */
+    size_t cbveclen   /* (i) Codebook vector length */
+                            ){
+  size_t k, base_size;
+  size_t lag;
+  /* Stack based */
+  int16_t tempbuff2[SUBL+5];
+
+  /* Determine size of codebook sections */
+
+  base_size=lMem-cbveclen+1;
+
+  if (cbveclen==SUBL) {
+    base_size += cbveclen / 2;
+  }
+
+  /* No filter -> First codebook section */
+
+  if (index<lMem-cbveclen+1) {
+
+    /* first non-interpolated vectors */
+
+    k=index+cbveclen;
+    /* get vector */
+    WEBRTC_SPL_MEMCPY_W16(cbvec, mem+lMem-k, cbveclen);
+
+  } else if (index < base_size) {
+
+    /* Calculate lag */
+
+    k = (2 * (index - (lMem - cbveclen + 1))) + cbveclen;
+
+    lag = k / 2;
+
+    WebRtcIlbcfix_CreateAugmentedVec(lag, mem+lMem, cbvec);
+
+  }
+
+  /* Higher codebbok section based on filtering */
+
+  else {
+
+    size_t memIndTest;
+
+    /* first non-interpolated vectors */
+
+    if (index-base_size<lMem-cbveclen+1) {
+
+      /* Set up filter memory, stuff zeros outside memory buffer */
+
+      memIndTest = lMem-(index-base_size+cbveclen);
+
+      WebRtcSpl_MemSetW16(mem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN);
+      WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN);
+
+      /* do filtering to get the codebook vector */
+
+      WebRtcSpl_FilterMAFastQ12(
+          &mem[memIndTest+4], cbvec, (int16_t*)WebRtcIlbcfix_kCbFiltersRev,
+          CB_FILTERLEN, cbveclen);
+    }
+
+    /* interpolated vectors */
+
+    else {
+      if (cbveclen < SUBL) {
+        // We're going to fill in cbveclen + 5 elements of tempbuff2 in
+        // WebRtcSpl_FilterMAFastQ12, less than the SUBL + 5 elements we'll be
+        // using in WebRtcIlbcfix_CreateAugmentedVec. This error is caused by
+        // bad values in `index` (which come from the encoded stream). Tell the
+        // caller that things went south, and that the decoder state is now
+        // corrupt (because it's half-way through an update that we can't
+        // complete).
+        return false;
+      }
+
+      /* Stuff zeros outside memory buffer  */
+      memIndTest = lMem-cbveclen-CB_FILTERLEN;
+      WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN);
+
+      /* do filtering */
+      WebRtcSpl_FilterMAFastQ12(
+          &mem[memIndTest+7], tempbuff2, (int16_t*)WebRtcIlbcfix_kCbFiltersRev,
+          CB_FILTERLEN, cbveclen+5);
+
+      /* Calculate lag index */
+      lag = (cbveclen<<1)-20+index-base_size-lMem-1;
+
+      WebRtcIlbcfix_CreateAugmentedVec(lag, tempbuff2+SUBL+5, cbvec);
+    }
+  }
+
+  return true;  // Success.
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
new file mode 100644
index 0000000000..99537dd0f7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetCbVec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
+
+#include <stdbool.h>
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/base/attributes.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+// Returns true on success, false on failure. In case of failure, the decoder
+// state may be corrupted and needs resetting.
+ABSL_MUST_USE_RESULT
+bool WebRtcIlbcfix_GetCbVec(
+    int16_t* cbvec, /* (o) Constructed codebook vector */
+    int16_t* mem,   /* (i) Codebook buffer */
+    size_t index,   /* (i) Codebook index */
+    size_t lMem,    /* (i) Length of codebook buffer */
+    size_t cbveclen /* (i) Codebook vector length */
+);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
new file mode 100644
index 0000000000..e0fb21caf0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetLspPoly.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/get_lsp_poly.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * Construct the polynomials F1(z) and F2(z) from the LSP
+ * (Computations are done in Q24)
+ *
+ * The expansion is performed using the following recursion:
+ *
+ * f[0] = 1;
+ * tmp = -2.0 * lsp[0];
+ * f[1] = tmp;
+ * for (i=2; i<=5; i++) {
+ *    b = -2.0 * lsp[2*i-2];
+ *    f[i] = tmp*f[i-1] + 2.0*f[i-2];
+ *    for (j=i; j>=2; j--) {
+ *       f[j] = f[j] + tmp*f[j-1] + f[j-2];
+ *    }
+ *    f[i] = f[i] + tmp;
+ * }
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetLspPoly(
+    int16_t *lsp, /* (i) LSP in Q15 */
+    int32_t *f)  /* (o) polonymial in Q24 */
+{
+  int32_t tmpW32;
+  int i, j;
+  int16_t high, low;
+  int16_t *lspPtr;
+  int32_t *fPtr;
+
+  lspPtr = lsp;
+  fPtr = f;
+  /* f[0] = 1.0 (Q24) */
+  (*fPtr) = (int32_t)16777216;
+  fPtr++;
+
+  (*fPtr) = WEBRTC_SPL_MUL((*lspPtr), -1024);
+  fPtr++;
+  lspPtr+=2;
+
+  for(i=2; i<=5; i++)
+  {
+    (*fPtr) = fPtr[-2];
+
+    for(j=i; j>1; j--)
+    {
+      /* Compute f[j] = f[j] + tmp*f[j-1] + f[j-2]; */
+      high = (int16_t)(fPtr[-1] >> 16);
+      low = (int16_t)((fPtr[-1] & 0xffff) >> 1);
+
+      tmpW32 = 4 * high * *lspPtr + 4 * ((low * *lspPtr) >> 15);
+
+      (*fPtr) += fPtr[-2];
+      (*fPtr) -= tmpW32;
+      fPtr--;
+    }
+    *fPtr -= *lspPtr * (1 << 10);
+
+    fPtr+=i;
+    lspPtr+=2;
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
new file mode 100644
index 0000000000..70c9c4d4b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetLspPoly.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * Construct the polynomials F1(z) and F2(z) from the LSP
+ * (Computations are done in Q24)
+ *
+ * The expansion is performed using the following recursion:
+ *
+ * f[0] = 1;
+ * tmp = -2.0 * lsp[0];
+ * f[1] = tmp;
+ * for (i=2; i<=5; i++) {
+ *    b = -2.0 * lsp[2*i-2];
+ *    f[i] = tmp*f[i-1] + 2.0*f[i-2];
+ *    for (j=i; j>=2; j--) {
+ *       f[j] = f[j] + tmp*f[j-1] + f[j-2];
+ *    }
+ *    f[i] = f[i] + tmp;
+ * }
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetLspPoly(int16_t* lsp, /* (i) LSP in Q15 */
+                              int32_t* f);  /* (o) polonymial in Q24 */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
new file mode 100644
index 0000000000..68a569a40a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetSyncSeq.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/get_sync_seq.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/nearest_neighbor.h"
+#include "modules/audio_coding/codecs/ilbc/refiner.h"
+
+/*----------------------------------------------------------------*
+ * get the pitch-synchronous sample sequence
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetSyncSeq(
+    int16_t *idata,   /* (i) original data */
+    size_t idatal,   /* (i) dimension of data */
+    size_t centerStartPos, /* (i) where current block starts */
+    size_t *period,   /* (i) rough-pitch-period array       (Q-2) */
+    const size_t *plocs, /* (i) where periods of period array are taken (Q-2) */
+    size_t periodl,   /* (i) dimension period array */
+    size_t hl,    /* (i) 2*hl+1 is the number of sequences */
+    int16_t *surround  /* (i/o) The contribution from this sequence
+                                summed with earlier contributions */
+                              ){
+  size_t i, centerEndPos, q;
+  /* Stack based */
+  size_t lagBlock[2 * ENH_HL + 1];
+  size_t blockStartPos[2 * ENH_HL + 1]; /* The position to search around (Q2) */
+  size_t plocs2[ENH_PLOCSL];
+
+  centerEndPos = centerStartPos + ENH_BLOCKL - 1;
+
+  /* present (find predicted lag from this position) */
+
+  WebRtcIlbcfix_NearestNeighbor(lagBlock + hl,
+                                plocs,
+                                2 * (centerStartPos + centerEndPos),
+                                periodl);
+
+  blockStartPos[hl] = 4 * centerStartPos;
+
+  /* past (find predicted position and perform a refined
+     search to find the best sequence) */
+
+  for (q = hl; q > 0; q--) {
+    size_t qq = q - 1;
+    size_t period_q = period[lagBlock[q]];
+    /* Stop if this sequence would be outside the buffer; that means all
+       further-past sequences would also be outside the buffer. */
+    if (blockStartPos[q] < period_q + (4 * ENH_OVERHANG))
+      break;
+    blockStartPos[qq] = blockStartPos[q] - period_q;
+
+    size_t value = blockStartPos[qq] + 4 * ENH_BLOCKL_HALF;
+    value = (value > period_q) ? (value - period_q) : 0;
+    WebRtcIlbcfix_NearestNeighbor(lagBlock + qq, plocs, value, periodl);
+
+    /* Find the best possible sequence in the 4 times upsampled
+        domain around blockStartPos+q */
+    WebRtcIlbcfix_Refiner(blockStartPos + qq, idata, idatal, centerStartPos,
+                          blockStartPos[qq], surround,
+                          WebRtcIlbcfix_kEnhWt[qq]);
+  }
+
+  /* future (find predicted position and perform a refined
+     search to find the best sequence) */
+
+  for (i = 0; i < periodl; i++) {
+    plocs2[i] = plocs[i] - period[i];
+  }
+
+  for (q = hl + 1; q <= (2 * hl); q++) {
+
+    WebRtcIlbcfix_NearestNeighbor(
+        lagBlock + q,
+        plocs2,
+        blockStartPos[q - 1] + 4 * ENH_BLOCKL_HALF,
+        periodl);
+
+    blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]];
+
+    if (blockStartPos[q] + 4 * (ENH_BLOCKL + ENH_OVERHANG) < 4 * idatal) {
+
+      /* Find the best possible sequence in the 4 times upsampled
+         domain around blockStartPos+q */
+      WebRtcIlbcfix_Refiner(blockStartPos + q, idata, idatal, centerStartPos,
+                            blockStartPos[q], surround,
+                            WebRtcIlbcfix_kEnhWt[2 * hl - q]);
+
+    } else {
+      /* Don't add anything since this sequence would
+         be outside the buffer */
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
new file mode 100644
index 0000000000..90962fa063
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_GetSyncSeq.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * get the pitch-synchronous sample sequence
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_GetSyncSeq(
+    int16_t* idata,        /* (i) original data */
+    size_t idatal,         /* (i) dimension of data */
+    size_t centerStartPos, /* (i) where current block starts */
+    size_t* period,        /* (i) rough-pitch-period array       (Q-2) */
+    const size_t* plocs, /* (i) where periods of period array are taken (Q-2) */
+    size_t periodl,      /* (i) dimension period array */
+    size_t hl,           /* (i) 2*hl+1 is the number of sequences */
+    int16_t* surround    /* (i/o) The contribution from this sequence
+                                  summed with earlier contributions */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.c
new file mode 100644
index 0000000000..be582f2e23
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.c
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpInput.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/hp_input.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  high-pass filter of input with *0.5 and saturation
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_HpInput(
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
+                                                                   {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+                                                                   is assumed to be 1.0 */
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                                                                   yhi[n-2] ylow[n-2] */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len)      /* (i)   Number of samples to filter */
+{
+  size_t i;
+  int32_t tmpW32;
+  int32_t tmpW32b;
+
+  for (i=0; i<len; i++) {
+
+    /*
+        y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
+        + (-a[1])*y[i-1] + (-a[2])*y[i-2];
+    */
+
+    tmpW32 = y[1] * ba[3];  /* (-a[1])*y[i-1] (low part) */
+    tmpW32 += y[3] * ba[4];  /* (-a[2])*y[i-2] (low part) */
+    tmpW32 = (tmpW32>>15);
+    tmpW32 += y[0] * ba[3];  /* (-a[1])*y[i-1] (high part) */
+    tmpW32 += y[2] * ba[4];  /* (-a[2])*y[i-2] (high part) */
+    tmpW32 = (tmpW32<<1);
+
+    tmpW32 += signal[i] * ba[0];  /* b[0]*x[0] */
+    tmpW32 += x[0] * ba[1];  /* b[1]*x[i-1] */
+    tmpW32 += x[1] * ba[2];  /* b[2]*x[i-2] */
+
+    /* Update state (input part) */
+    x[1] = x[0];
+    x[0] = signal[i];
+
+    /* Rounding in Q(12+1), i.e. add 2^12 */
+    tmpW32b = tmpW32 + 4096;
+
+    /* Saturate (to 2^28) so that the HP filtered signal does not overflow */
+    tmpW32b = WEBRTC_SPL_SAT((int32_t)268435455, tmpW32b, (int32_t)-268435456);
+
+    /* Convert back to Q0 and multiply with 0.5 */
+    signal[i] = (int16_t)(tmpW32b >> 13);
+
+    /* Update state (filtered part) */
+    y[2] = y[0];
+    y[3] = y[1];
+
+    /* upshift tmpW32 by 3 with saturation */
+    if (tmpW32>268435455) {
+      tmpW32 = WEBRTC_SPL_WORD32_MAX;
+    } else if (tmpW32<-268435456) {
+      tmpW32 = WEBRTC_SPL_WORD32_MIN;
+    } else {
+      tmpW32 <<= 3;
+    }
+
+    y[0] = (int16_t)(tmpW32 >> 16);
+    y[1] = (int16_t)((tmpW32 - (y[0] << 16)) >> 1);
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.h
new file mode 100644
index 0000000000..9143d8efed
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_input.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpInput.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+// clang-format off
+// Bad job here. https://bugs.llvm.org/show_bug.cgi?id=34274
+void WebRtcIlbcfix_HpInput(
+    int16_t* signal, /* (i/o) signal vector */
+    int16_t* ba,     /* (i)   B- and A-coefficients (2:nd order)
+                              {b[0] b[1] b[2] -a[1] -a[2]}
+                              a[0] is assumed to be 1.0 */
+    int16_t* y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                              yhi[n-2] ylow[n-2] */
+    int16_t* x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len);     /* (i)   Number of samples to filter */
+// clang-format on
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.c
new file mode 100644
index 0000000000..cc5f6dcd37
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.c
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpOutput.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  high-pass filter of output and *2 with saturation
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_HpOutput(
+    int16_t *signal,     /* (i/o) signal vector */
+    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
+                                                                   {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+                                                                   is assumed to be 1.0 */
+    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                                                                   yhi[n-2] ylow[n-2] */
+    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len)      /* (i)   Number of samples to filter */
+{
+  size_t i;
+  int32_t tmpW32;
+  int32_t tmpW32b;
+
+  for (i=0; i<len; i++) {
+
+    /*
+      y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
+      + (-a[1])*y[i-1] + (-a[2])*y[i-2];
+    */
+
+    tmpW32 = y[1] * ba[3];  /* (-a[1])*y[i-1] (low part) */
+    tmpW32 += y[3] * ba[4];  /* (-a[2])*y[i-2] (low part) */
+    tmpW32 = (tmpW32>>15);
+    tmpW32 += y[0] * ba[3];  /* (-a[1])*y[i-1] (high part) */
+    tmpW32 += y[2] * ba[4];  /* (-a[2])*y[i-2] (high part) */
+    tmpW32 *= 2;
+
+    tmpW32 += signal[i] * ba[0];  /* b[0]*x[0] */
+    tmpW32 += x[0] * ba[1];  /* b[1]*x[i-1] */
+    tmpW32 += x[1] * ba[2];  /* b[2]*x[i-2] */
+
+    /* Update state (input part) */
+    x[1] = x[0];
+    x[0] = signal[i];
+
+    /* Rounding in Q(12-1), i.e. add 2^10 */
+    tmpW32b = tmpW32 + 1024;
+
+    /* Saturate (to 2^26) so that the HP filtered signal does not overflow */
+    tmpW32b = WEBRTC_SPL_SAT((int32_t)67108863, tmpW32b, (int32_t)-67108864);
+
+    /* Convert back to Q0 and multiply with 2 */
+    signal[i] = (int16_t)(tmpW32b >> 11);
+
+    /* Update state (filtered part) */
+    y[2] = y[0];
+    y[3] = y[1];
+
+    /* upshift tmpW32 by 3 with saturation */
+    if (tmpW32>268435455) {
+      tmpW32 = WEBRTC_SPL_WORD32_MAX;
+    } else if (tmpW32<-268435456) {
+      tmpW32 = WEBRTC_SPL_WORD32_MIN;
+    } else {
+      tmpW32 *= 8;
+    }
+
+    y[0] = (int16_t)(tmpW32 >> 16);
+    y[1] = (int16_t)((tmpW32 & 0xffff) >> 1);
+
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.h
new file mode 100644
index 0000000000..6d1bd3cd88
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/hp_output.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_HpOutput.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+// clang-format off
+// Bad job here. https://bugs.llvm.org/show_bug.cgi?id=34274
+void WebRtcIlbcfix_HpOutput(
+    int16_t* signal, /* (i/o) signal vector */
+    int16_t* ba,     /* (i)   B- and A-coefficients (2:nd order)
+                              {b[0] b[1] b[2] -a[1] -a[2]} a[0]
+                              is assumed to be 1.0 */
+    int16_t* y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
+                              yhi[n-2] ylow[n-2] */
+    int16_t* x,      /* (i/o) Filter state x[n-1] x[n-2] */
+    size_t len);     /* (i)   Number of samples to filter */
+// clang-format on
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.c
new file mode 100644
index 0000000000..ba6c3e46c3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -0,0 +1,288 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ iLBCInterface.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+#include <stdlib.h>
+
+#include "modules/audio_coding/codecs/ilbc/decode.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/encode.h"
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "modules/audio_coding/codecs/ilbc/init_encode.h"
+#include "rtc_base/checks.h"
+
+int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst,
+                                    int16_t* ILBCENC_inst_Addr,
+                                    int16_t* size) {
+  *iLBC_encinst=(IlbcEncoderInstance*)ILBCENC_inst_Addr;
+  *size=sizeof(IlbcEncoder)/sizeof(int16_t);
+  if (*iLBC_encinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_DecoderAssign(IlbcDecoderInstance** iLBC_decinst,
+                                    int16_t* ILBCDEC_inst_Addr,
+                                    int16_t* size) {
+  *iLBC_decinst=(IlbcDecoderInstance*)ILBCDEC_inst_Addr;
+  *size=sizeof(IlbcDecoder)/sizeof(int16_t);
+  if (*iLBC_decinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance **iLBC_encinst) {
+  *iLBC_encinst=(IlbcEncoderInstance*)malloc(sizeof(IlbcEncoder));
+  if (*iLBC_encinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance **iLBC_decinst) {
+  *iLBC_decinst=(IlbcDecoderInstance*)malloc(sizeof(IlbcDecoder));
+  if (*iLBC_decinst!=NULL) {
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int16_t WebRtcIlbcfix_EncoderFree(IlbcEncoderInstance *iLBC_encinst) {
+  free(iLBC_encinst);
+  return(0);
+}
+
+int16_t WebRtcIlbcfix_DecoderFree(IlbcDecoderInstance *iLBC_decinst) {
+  free(iLBC_decinst);
+  return(0);
+}
+
+int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
+                                  int16_t mode) {
+  if ((mode==20)||(mode==30)) {
+    WebRtcIlbcfix_InitEncode((IlbcEncoder*) iLBCenc_inst, mode);
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+
+int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded) {
+  size_t pos = 0;
+  size_t encpos = 0;
+
+  if ((len != ((IlbcEncoder*)iLBCenc_inst)->blockl) &&
+#ifdef SPLIT_10MS
+      (len != 80) &&
+#endif
+      (len != 2*((IlbcEncoder*)iLBCenc_inst)->blockl) &&
+      (len != 3*((IlbcEncoder*)iLBCenc_inst)->blockl))
+  {
+    /* A maximum of 3 frames/packet is allowed */
+    return(-1);
+  } else {
+
+    /* call encoder */
+    while (pos<len) {
+      WebRtcIlbcfix_EncodeImpl((uint16_t*)&encoded[2 * encpos], &speechIn[pos],
+                               (IlbcEncoder*)iLBCenc_inst);
+#ifdef SPLIT_10MS
+      pos += 80;
+      if(((IlbcEncoder*)iLBCenc_inst)->section == 0)
+#else
+        pos += ((IlbcEncoder*)iLBCenc_inst)->blockl;
+#endif
+      encpos += ((IlbcEncoder*)iLBCenc_inst)->no_of_words;
+    }
+    return (int)(encpos*2);
+  }
+}
+
+int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
+                                  int16_t mode) {
+  if ((mode==20)||(mode==30)) {
+    WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, mode, 1);
+    return(0);
+  } else {
+    return(-1);
+  }
+}
+void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst) {
+  WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 20, 1);
+}
+void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst) {
+  WebRtcIlbcfix_InitDecode((IlbcDecoder*) iLBCdec_inst, 30, 1);
+}
+
+
+int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType)
+{
+  size_t i=0;
+  /* Allow for automatic switching between the frame sizes
+     (although you do get some discontinuity) */
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
+    /* ok, do nothing */
+  } else {
+    /* Test if the mode has changed */
+    if (((IlbcDecoder*)iLBCdec_inst)->mode==20) {
+      if ((len==NO_OF_BYTES_30MS)||
+          (len==2*NO_OF_BYTES_30MS)||
+          (len==3*NO_OF_BYTES_30MS)) {
+        WebRtcIlbcfix_InitDecode(
+            ((IlbcDecoder*)iLBCdec_inst), 30,
+            ((IlbcDecoder*)iLBCdec_inst)->use_enhancer);
+      } else {
+        /* Unsupported frame length */
+        return(-1);
+      }
+    } else {
+      if ((len==NO_OF_BYTES_20MS)||
+          (len==2*NO_OF_BYTES_20MS)||
+          (len==3*NO_OF_BYTES_20MS)) {
+        WebRtcIlbcfix_InitDecode(
+            ((IlbcDecoder*)iLBCdec_inst), 20,
+            ((IlbcDecoder*)iLBCdec_inst)->use_enhancer);
+      } else {
+        /* Unsupported frame length */
+        return(-1);
+      }
+    }
+  }
+
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
+    if (WebRtcIlbcfix_DecodeImpl(
+            &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
+            (const uint16_t*)&encoded
+                [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+            (IlbcDecoder*)iLBCdec_inst, 1) == -1)
+      return -1;
+    i++;
+  }
+  /* iLBC does not support VAD/CNG yet */
+  *speechType=1;
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType)
+{
+  size_t i=0;
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
+    /* ok, do nothing */
+  } else {
+    return(-1);
+  }
+
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
+    if (!WebRtcIlbcfix_DecodeImpl(
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
+        (const uint16_t*)&encoded
+            [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+        (IlbcDecoder*)iLBCdec_inst, 1))
+      return -1;
+    i++;
+  }
+  /* iLBC does not support VAD/CNG yet */
+  *speechType=1;
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType)
+{
+  size_t i=0;
+  if ((len==((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==2*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)||
+      (len==3*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)) {
+    /* ok, do nothing */
+  } else {
+    return(-1);
+  }
+
+  while ((i*((IlbcDecoder*)iLBCdec_inst)->no_of_bytes)<len) {
+    if (!WebRtcIlbcfix_DecodeImpl(
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl],
+        (const uint16_t*)&encoded
+            [2 * i * ((IlbcDecoder*)iLBCdec_inst)->no_of_words],
+        (IlbcDecoder*)iLBCdec_inst, 1))
+      return -1;
+    i++;
+  }
+  /* iLBC does not support VAD/CNG yet */
+  *speechType=1;
+  return (int)(i*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
+                               int16_t* decoded,
+                               size_t noOfLostFrames) {
+  size_t i;
+  uint16_t dummy;
+
+  for (i=0;i<noOfLostFrames;i++) {
+    // PLC decoding shouldn't fail, because there is no external input data
+    // that can be bad.
+    int result = WebRtcIlbcfix_DecodeImpl(
+        &decoded[i * ((IlbcDecoder*)iLBCdec_inst)->blockl], &dummy,
+        (IlbcDecoder*)iLBCdec_inst, 0);
+    RTC_CHECK_EQ(result, 0);
+  }
+  return (noOfLostFrames*((IlbcDecoder*)iLBCdec_inst)->blockl);
+}
+
+size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst,
+                              int16_t* decoded,
+                              size_t noOfLostFrames) {
+  /* Two input parameters not used, but needed for function pointers in NetEQ */
+  (void)(decoded = NULL);
+  (void)(noOfLostFrames = 0);
+
+  WebRtcSpl_MemSetW16(((IlbcDecoder*)iLBCdec_inst)->enh_buf, 0, ENH_BUFL);
+  ((IlbcDecoder*)iLBCdec_inst)->prev_enh_pl = 2;
+
+  return (0);
+}
+
+void WebRtcIlbcfix_version(char *version)
+{
+  strcpy((char*)version, "1.1.1");
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.h
new file mode 100644
index 0000000000..de8cfde111
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc.h
@@ -0,0 +1,251 @@
+/*
+ *  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.
+ */
+
+/*
+ * ilbc.h
+ *
+ * This header file contains all of the API's for iLBC.
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*
+ * Solution to support multiple instances
+ * Customer has to cast instance to proper type
+ */
+
+typedef struct iLBC_encinst_t_ IlbcEncoderInstance;
+
+typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
+
+/*
+ * Comfort noise constants
+ */
+
+#define ILBC_SPEECH 1
+#define ILBC_CNG 2
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcIlbcfix_XxxAssign(...)
+ *
+ * These functions assigns the encoder/decoder instance to the specified
+ * memory location
+ *
+ * Input:
+ *     - XXX_xxxinst       : Pointer to created instance that should be
+ *                           assigned
+ *     - ILBCXXX_inst_Addr : Pointer to the desired memory space
+ *     - size              : The size that this structure occupies (in Word16)
+ *
+ * Return value             :  0 - Ok
+ *                            -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst,
+                                    int16_t* ILBCENC_inst_Addr,
+                                    int16_t* size);
+int16_t WebRtcIlbcfix_DecoderAssign(IlbcDecoderInstance** iLBC_decinst,
+                                    int16_t* ILBCDEC_inst_Addr,
+                                    int16_t* size);
+
+/****************************************************************************
+ * WebRtcIlbcfix_XxxAssign(...)
+ *
+ * These functions create a instance to the specified structure
+ *
+ * Input:
+ *      - XXX_inst        : Pointer to created instance that should be created
+ *
+ * Return value           :  0 - Ok
+ *                          -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance** iLBC_encinst);
+int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance** iLBC_decinst);
+
+/****************************************************************************
+ * WebRtcIlbcfix_XxxFree(...)
+ *
+ * These functions frees the dynamic memory of a specified instance
+ *
+ * Input:
+ *      - XXX_inst          : Pointer to created instance that should be freed
+ *
+ * Return value             :  0 - Ok
+ *                            -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderFree(IlbcEncoderInstance* iLBC_encinst);
+int16_t WebRtcIlbcfix_DecoderFree(IlbcDecoderInstance* iLBC_decinst);
+
+/****************************************************************************
+ * WebRtcIlbcfix_EncoderInit(...)
+ *
+ * This function initializes a iLBC instance
+ *
+ * Input:
+ *      - iLBCenc_inst      : iLBC instance, i.e. the user that should receive
+ *                            be initialized
+ *      - frameLen          : The frame length of the codec 20/30 (ms)
+ *
+ * Return value             :  0 - Ok
+ *                            -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
+                                  int16_t frameLen);
+
+/****************************************************************************
+ * WebRtcIlbcfix_Encode(...)
+ *
+ * This function encodes one iLBC frame. Input speech length has be a
+ * multiple of the frame length.
+ *
+ * Input:
+ *      - iLBCenc_inst      : iLBC instance, i.e. the user that should encode
+ *                            a package
+ *      - speechIn          : Input speech vector
+ *      - len               : Samples in speechIn (160, 240, 320 or 480)
+ *
+ * Output:
+ *  - encoded               : The encoded data vector
+ *
+ * Return value             : >0 - Length (in bytes) of coded data
+ *                            -1 - Error
+ */
+
+int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
+                         const int16_t* speechIn,
+                         size_t len,
+                         uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcIlbcfix_DecoderInit(...)
+ *
+ * This function initializes a iLBC instance with either 20 or 30 ms frames
+ * Alternatively the WebRtcIlbcfix_DecoderInit_XXms can be used. Then it's
+ * not needed to specify the frame length with a variable.
+ *
+ * Input:
+ *      - IlbcDecoderInstance : iLBC decoder instance
+ *      - frameLen            : The frame length of the codec 20/30 (ms)
+ *
+ * Return value               :  0 - Ok
+ *                              -1 - Error
+ */
+
+int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
+                                  int16_t frameLen);
+void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst);
+void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst);
+
+/****************************************************************************
+ * WebRtcIlbcfix_Decode(...)
+ *
+ * This function decodes a packet with iLBC frame(s). Output speech length
+ * will be a multiple of 160 or 240 samples ((160 or 240)*frames/packet).
+ *
+ * Input:
+ *      - iLBCdec_inst      : iLBC instance, i.e. the user that should decode
+ *                            a packet
+ *      - encoded           : Encoded iLBC frame(s)
+ *      - len               : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded           : The decoded vector
+ *      - speechType        : 1 normal, 2 CNG
+ *
+ * Return value             : >0 - Samples in decoded vector
+ *                            -1 - Error
+ */
+
+int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
+                         const uint8_t* encoded,
+                         size_t len,
+                         int16_t* decoded,
+                         int16_t* speechType);
+int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType);
+int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
+                             const uint8_t* encoded,
+                             size_t len,
+                             int16_t* decoded,
+                             int16_t* speechType);
+
+/****************************************************************************
+ * WebRtcIlbcfix_DecodePlc(...)
+ *
+ * This function conducts PLC for iLBC frame(s). Output speech length
+ * will be a multiple of 160 or 240 samples.
+ *
+ * Input:
+ *      - iLBCdec_inst      : iLBC instance, i.e. the user that should perform
+ *                            a PLC
+ *      - noOfLostFrames    : Number of PLC frames to produce
+ *
+ * Output:
+ *      - decoded           : The "decoded" vector
+ *
+ * Return value             : Samples in decoded PLC vector
+ */
+
+size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
+                               int16_t* decoded,
+                               size_t noOfLostFrames);
+
+/****************************************************************************
+ * WebRtcIlbcfix_NetEqPlc(...)
+ *
+ * This function updates the decoder when a packet loss has occured, but it
+ * does not produce any PLC data. Function can be used if another PLC method
+ * is used (i.e NetEq).
+ *
+ * Input:
+ *      - iLBCdec_inst      : iLBC instance that should be updated
+ *      - noOfLostFrames    : Number of lost frames
+ *
+ * Output:
+ *      - decoded           : The "decoded" vector (nothing in this case)
+ *
+ * Return value             : Samples in decoded PLC vector
+ */
+
+size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst,
+                              int16_t* decoded,
+                              size_t noOfLostFrames);
+
+/****************************************************************************
+ * WebRtcIlbcfix_version(...)
+ *
+ * This function returns the version number of iLBC
+ *
+ * Output:
+ *      - version           : Version number of iLBC (maximum 20 char)
+ */
+
+void WebRtcIlbcfix_version(char* version);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
new file mode 100644
index 0000000000..689292f131
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
@@ -0,0 +1,140 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(IlbcTest, BadPacket) {
+  // Get a good packet.
+  AudioEncoderIlbcConfig config;
+  config.frame_size_ms = 20;  // We need 20 ms rather than the default 30 ms;
+                              // otherwise, all possible values of cb_index[2]
+                              // are valid.
+  AudioEncoderIlbcImpl encoder(config, 102);
+  std::vector<int16_t> samples(encoder.SampleRateHz() / 100, 4711);
+  rtc::Buffer packet;
+  int num_10ms_chunks = 0;
+  while (packet.size() == 0) {
+    encoder.Encode(0, samples, &packet);
+    num_10ms_chunks += 1;
+  }
+
+  // Break the packet by setting all bits of the unsigned 7-bit number
+  // cb_index[2] to 1, giving it a value of 127. For a 20 ms packet, this is
+  // too large.
+  EXPECT_EQ(38u, packet.size());
+  rtc::Buffer bad_packet(packet.data(), packet.size());
+  bad_packet[29] |= 0x3f;  // Bits 1-6.
+  bad_packet[30] |= 0x80;  // Bit 0.
+
+  // Decode the bad packet. We expect the decoder to respond by returning -1.
+  AudioDecoderIlbcImpl decoder;
+  std::vector<int16_t> decoded_samples(num_10ms_chunks * samples.size());
+  AudioDecoder::SpeechType speech_type;
+  EXPECT_EQ(-1, decoder.Decode(bad_packet.data(), bad_packet.size(),
+                               encoder.SampleRateHz(),
+                               sizeof(int16_t) * decoded_samples.size(),
+                               decoded_samples.data(), &speech_type));
+
+  // Decode the good packet. This should work, because the failed decoding
+  // should not have left the decoder in a broken state.
+  EXPECT_EQ(static_cast<int>(decoded_samples.size()),
+            decoder.Decode(packet.data(), packet.size(), encoder.SampleRateHz(),
+                           sizeof(int16_t) * decoded_samples.size(),
+                           decoded_samples.data(), &speech_type));
+}
+
+class SplitIlbcTest : public ::testing::TestWithParam<std::pair<int, int> > {
+ protected:
+  virtual void SetUp() {
+    const std::pair<int, int> parameters = GetParam();
+    num_frames_ = parameters.first;
+    frame_length_ms_ = parameters.second;
+    frame_length_bytes_ = (frame_length_ms_ == 20) ? 38 : 50;
+  }
+  size_t num_frames_;
+  int frame_length_ms_;
+  size_t frame_length_bytes_;
+};
+
+TEST_P(SplitIlbcTest, NumFrames) {
+  AudioDecoderIlbcImpl decoder;
+  const size_t frame_length_samples = frame_length_ms_ * 8;
+  const auto generate_payload = [](size_t payload_length_bytes) {
+    rtc::Buffer payload(payload_length_bytes);
+    // Fill payload with increasing integers {0, 1, 2, ...}.
+    for (size_t i = 0; i < payload.size(); ++i) {
+      payload[i] = static_cast<uint8_t>(i);
+    }
+    return payload;
+  };
+
+  const auto results = decoder.ParsePayload(
+      generate_payload(frame_length_bytes_ * num_frames_), 0);
+  EXPECT_EQ(num_frames_, results.size());
+
+  size_t frame_num = 0;
+  uint8_t payload_value = 0;
+  for (const auto& result : results) {
+    EXPECT_EQ(frame_length_samples * frame_num, result.timestamp);
+    const LegacyEncodedAudioFrame* frame =
+        static_cast<const LegacyEncodedAudioFrame*>(result.frame.get());
+    const rtc::Buffer& payload = frame->payload();
+    EXPECT_EQ(frame_length_bytes_, payload.size());
+    for (size_t i = 0; i < payload.size(); ++i, ++payload_value) {
+      EXPECT_EQ(payload_value, payload[i]);
+    }
+    ++frame_num;
+  }
+}
+
+// Test 1 through 5 frames of 20 and 30 ms size.
+// Also test the maximum number of frames in one packet for 20 and 30 ms.
+// The maximum is defined by the largest payload length that can be uniquely
+// resolved to a frame size of either 38 bytes (20 ms) or 50 bytes (30 ms).
+INSTANTIATE_TEST_SUITE_P(
+    IlbcTest,
+    SplitIlbcTest,
+    ::testing::Values(std::pair<int, int>(1, 20),  // 1 frame, 20 ms.
+                      std::pair<int, int>(2, 20),  // 2 frames, 20 ms.
+                      std::pair<int, int>(3, 20),  // And so on.
+                      std::pair<int, int>(4, 20),
+                      std::pair<int, int>(5, 20),
+                      std::pair<int, int>(24, 20),
+                      std::pair<int, int>(1, 30),
+                      std::pair<int, int>(2, 30),
+                      std::pair<int, int>(3, 30),
+                      std::pair<int, int>(4, 30),
+                      std::pair<int, int>(5, 30),
+                      std::pair<int, int>(18, 30)));
+
+// Test too large payload size.
+TEST(IlbcTest, SplitTooLargePayload) {
+  AudioDecoderIlbcImpl decoder;
+  constexpr size_t kPayloadLengthBytes = 950;
+  const auto results =
+      decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0);
+  EXPECT_TRUE(results.empty());
+}
+
+// Payload not an integer number of frames.
+TEST(IlbcTest, SplitUnevenPayload) {
+  AudioDecoderIlbcImpl decoder;
+  constexpr size_t kPayloadLengthBytes = 39;  // Not an even number of frames.
+  const auto results =
+      decoder.ParsePayload(rtc::Buffer(kPayloadLengthBytes), 0);
+  EXPECT_TRUE(results.empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c
new file mode 100644
index 0000000000..d78f81a897
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvDec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_IndexConvDec(
+    int16_t *index   /* (i/o) Codebook indexes */
+                                ){
+  int k;
+
+  for (k=4;k<6;k++) {
+    /* Readjust the second and third codebook index for the first 40 sample
+       so that they look the same as the first (in terms of lag)
+    */
+    if ((index[k]>=44)&&(index[k]<108)) {
+      index[k]+=64;
+    } else if ((index[k]>=108)&&(index[k]<128)) {
+      index[k]+=128;
+    } else {
+      /* ERROR */
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h
new file mode 100644
index 0000000000..4f08ce04df
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvDec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_IndexConvDec(int16_t* index /* (i/o) Codebook indexes */
+                                );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c
new file mode 100644
index 0000000000..83144150b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ IiLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvEnc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/index_conv_enc.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Convert the codebook indexes to make the search easier
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_IndexConvEnc(
+    int16_t *index   /* (i/o) Codebook indexes */
+                                ){
+  int k;
+
+  for (k=4;k<6;k++) {
+    /* Readjust the second and third codebook index so that it is
+       packetized into 7 bits (before it was put in lag-wise the same
+       way as for the first codebook which uses 8 bits)
+    */
+    if ((index[k]>=108)&&(index[k]<172)) {
+      index[k]-=64;
+    } else if (index[k]>=236) {
+      index[k]-=128;
+    } else {
+      /* ERROR */
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h
new file mode 100644
index 0000000000..4fbf98084e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_IndexConvEnc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Convert the codebook indexes to make the search easier
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_IndexConvEnc(int16_t* index /* (i/o) Codebook indexes */
+                                );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.c
new file mode 100644
index 0000000000..3eb41e33b0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.c
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+	iLBC Speech Coder ANSI-C Source Code
+
+	WebRtcIlbcfix_InitDecode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of decoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
+    IlbcDecoder *iLBCdec_inst,  /* (i/o) Decoder instance */
+    int16_t mode,  /* (i) frame size mode */
+    int use_enhancer) {  /* (i) 1: use enhancer, 0: no enhancer */
+  int i;
+
+  iLBCdec_inst->mode = mode;
+
+  /* Set all the variables that are dependent on the frame size mode */
+  if (mode==30) {
+    iLBCdec_inst->blockl = BLOCKL_30MS;
+    iLBCdec_inst->nsub = NSUB_30MS;
+    iLBCdec_inst->nasub = NASUB_30MS;
+    iLBCdec_inst->lpc_n = LPC_N_30MS;
+    iLBCdec_inst->no_of_bytes = NO_OF_BYTES_30MS;
+    iLBCdec_inst->no_of_words = NO_OF_WORDS_30MS;
+    iLBCdec_inst->state_short_len=STATE_SHORT_LEN_30MS;
+  }
+  else if (mode==20) {
+    iLBCdec_inst->blockl = BLOCKL_20MS;
+    iLBCdec_inst->nsub = NSUB_20MS;
+    iLBCdec_inst->nasub = NASUB_20MS;
+    iLBCdec_inst->lpc_n = LPC_N_20MS;
+    iLBCdec_inst->no_of_bytes = NO_OF_BYTES_20MS;
+    iLBCdec_inst->no_of_words = NO_OF_WORDS_20MS;
+    iLBCdec_inst->state_short_len=STATE_SHORT_LEN_20MS;
+  }
+  else {
+    return(-1);
+  }
+
+  /* Reset all the previous LSF to mean LSF */
+  WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+
+  /* Clear the synthesis filter memory */
+  WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER);
+
+  /* Set the old synthesis filter to {1.0 0.0 ... 0.0} */
+  WebRtcSpl_MemSetW16(iLBCdec_inst->old_syntdenum, 0, ((LPC_FILTERORDER + 1)*NSUB_MAX));
+  for (i=0; i<NSUB_MAX; i++) {
+    iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)] = 4096;
+  }
+
+  /* Clear the variables that are used for the PLC */
+  iLBCdec_inst->last_lag = 20;
+  iLBCdec_inst->consPLICount = 0;
+  iLBCdec_inst->prevPLI = 0;
+  iLBCdec_inst->perSquare = 0;
+  iLBCdec_inst->prevLag = 120;
+  iLBCdec_inst->prevLpc[0] = 4096;
+  WebRtcSpl_MemSetW16(iLBCdec_inst->prevLpc+1, 0, LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(iLBCdec_inst->prevResidual, 0, BLOCKL_MAX);
+
+  /* Initialize the seed for the random number generator */
+  iLBCdec_inst->seed = 777;
+
+  /* Set the filter state of the HP filter to 0 */
+  WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2);
+  WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4);
+
+  /* Set the variables that are used in the ehnahcer */
+  iLBCdec_inst->use_enhancer = use_enhancer;
+  WebRtcSpl_MemSetW16(iLBCdec_inst->enh_buf, 0, (ENH_BUFL+ENH_BUFL_FILTEROVERHEAD));
+  for (i=0;i<ENH_NBLOCKS_TOT;i++) {
+    iLBCdec_inst->enh_period[i]=160; /* Q(-4) */
+  }
+
+  iLBCdec_inst->prev_enh_pl = 0;
+
+  return (int)(iLBCdec_inst->blockl);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.h
new file mode 100644
index 0000000000..a2b7b91287
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_decode.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitDecode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of decoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitDecode(/* (o) Number of decoded samples */
+                             IlbcDecoder*
+                                 iLBCdec_inst, /* (i/o) Decoder instance */
+                             int16_t mode,     /* (i) frame size mode */
+                             int use_enhancer  /* (i) 1 to use enhancer
+                                                  0 to run without enhancer */
+                             );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.c
new file mode 100644
index 0000000000..aa858e94bb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.c
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitEncode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/init_encode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of encoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
+    IlbcEncoder *iLBCenc_inst,  /* (i/o) Encoder instance */
+    int16_t mode) {  /* (i) frame size mode */
+  iLBCenc_inst->mode = mode;
+
+  /* Set all the variables that are dependent on the frame size mode */
+  if (mode==30) {
+    iLBCenc_inst->blockl = BLOCKL_30MS;
+    iLBCenc_inst->nsub = NSUB_30MS;
+    iLBCenc_inst->nasub = NASUB_30MS;
+    iLBCenc_inst->lpc_n = LPC_N_30MS;
+    iLBCenc_inst->no_of_bytes = NO_OF_BYTES_30MS;
+    iLBCenc_inst->no_of_words = NO_OF_WORDS_30MS;
+    iLBCenc_inst->state_short_len=STATE_SHORT_LEN_30MS;
+  }
+  else if (mode==20) {
+    iLBCenc_inst->blockl = BLOCKL_20MS;
+    iLBCenc_inst->nsub = NSUB_20MS;
+    iLBCenc_inst->nasub = NASUB_20MS;
+    iLBCenc_inst->lpc_n = LPC_N_20MS;
+    iLBCenc_inst->no_of_bytes = NO_OF_BYTES_20MS;
+    iLBCenc_inst->no_of_words = NO_OF_WORDS_20MS;
+    iLBCenc_inst->state_short_len=STATE_SHORT_LEN_20MS;
+  }
+  else {
+    return(-1);
+  }
+
+  /* Clear the buffers and set the previous LSF and LSP to the mean value */
+  WebRtcSpl_MemSetW16(iLBCenc_inst->anaMem, 0, LPC_FILTERORDER);
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(iLBCenc_inst->lpc_buffer, 0, LPC_LOOKBACK + BLOCKL_MAX);
+
+  /* Set the filter state of the HP filter to 0 */
+  WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemx, 0, 2);
+  WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemy, 0, 4);
+
+#ifdef SPLIT_10MS
+  /*Zeroing the past samples for 10msec Split*/
+  WebRtcSpl_MemSetW16(iLBCenc_inst->past_samples,0,160);
+  iLBCenc_inst->section = 0;
+#endif
+
+  return (int)(iLBCenc_inst->no_of_bytes);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.h
new file mode 100644
index 0000000000..4ada6a30c8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/init_encode.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InitEncode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  Initiation of encoder instance.
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_InitEncode(/* (o) Number of bytes encoded */
+                             IlbcEncoder*
+                                 iLBCenc_inst, /* (i/o) Encoder instance */
+                             int16_t mode      /* (i) frame size mode */
+                             );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.c
new file mode 100644
index 0000000000..17ed244bd4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.c
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Interpolate.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  interpolation between vectors
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Interpolate(
+    int16_t *out, /* (o) output vector */
+    int16_t *in1, /* (i) first input vector */
+    int16_t *in2, /* (i) second input vector */
+    int16_t coef, /* (i) weight coefficient in Q14 */
+    int16_t length)  /* (i) number of sample is vectors */
+{
+  int i;
+  int16_t invcoef;
+
+  /*
+    Performs the operation out[i] = in[i]*coef + (1-coef)*in2[i] (with rounding)
+  */
+
+  invcoef = 16384 - coef; /* 16384 = 1.0 (Q14)*/
+  for (i = 0; i < length; i++) {
+    out[i] = (int16_t)((coef * in1[i] + invcoef * in2[i] + 8192) >> 14);
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.h
new file mode 100644
index 0000000000..892082b75c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Interpolate.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  interpolation between vectors
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Interpolate(
+    int16_t* out,    /* (o) output vector */
+    int16_t* in1,    /* (i) first input vector */
+    int16_t* in2,    /* (i) second input vector */
+    int16_t coef,    /* (i) weight coefficient in Q14 */
+    int16_t length); /* (i) number of sample is vectors */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
new file mode 100644
index 0000000000..6dddd6fb86
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InterpolateSamples.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/interpolate_samples.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+void WebRtcIlbcfix_InterpolateSamples(
+    int16_t *interpSamples, /* (o) The interpolated samples */
+    int16_t *CBmem,   /* (i) The CB memory */
+    size_t lMem    /* (i) Length of the CB memory */
+                                      ) {
+  int16_t *ppi, *ppo, i, j, temp1, temp2;
+  int16_t *tmpPtr;
+
+  /* Calculate the 20 vectors of interpolated samples (4 samples each)
+     that are used in the codebooks for lag 20 to 39 */
+  tmpPtr = interpSamples;
+  for (j=0; j<20; j++) {
+    temp1 = 0;
+    temp2 = 3;
+    ppo = CBmem+lMem-4;
+    ppi = CBmem+lMem-j-24;
+    for (i=0; i<4; i++) {
+
+      *tmpPtr++ = (int16_t)((WebRtcIlbcfix_kAlpha[temp2] * *ppo) >> 15) +
+          (int16_t)((WebRtcIlbcfix_kAlpha[temp1] * *ppi) >> 15);
+
+      ppo++;
+      ppi++;
+      temp1++;
+      temp2--;
+    }
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
new file mode 100644
index 0000000000..bc665d7854
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_InterpolateSamples.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Construct the interpolated samples for the Augmented CB
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_InterpolateSamples(
+    int16_t* interpSamples, /* (o) The interpolated samples */
+    int16_t* CBmem,         /* (i) The CB memory */
+    size_t lMem             /* (i) Length of the CB memory */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
new file mode 100644
index 0000000000..89f6d29724
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LpcEncode.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lpc_encode.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+#include "modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h"
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_quant.h"
+
+/*----------------------------------------------------------------*
+ *  lpc encoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LpcEncode(
+    int16_t *syntdenum,  /* (i/o) synthesis filter coefficients
+                                           before/after encoding */
+    int16_t *weightdenum, /* (i/o) weighting denumerator coefficients
+                                   before/after encoding */
+    int16_t *lsf_index,  /* (o) lsf quantization index */
+    int16_t *data,   /* (i) Speech to do LPC analysis on */
+    IlbcEncoder *iLBCenc_inst
+    /* (i/o) the encoder state structure */
+                              ) {
+  /* Stack based */
+  int16_t lsf[LPC_FILTERORDER * LPC_N_MAX];
+  int16_t lsfdeq[LPC_FILTERORDER * LPC_N_MAX];
+
+  /* Calculate LSF's from the input speech */
+  WebRtcIlbcfix_SimpleLpcAnalysis(lsf, data, iLBCenc_inst);
+
+  /* Quantize the LSF's */
+  WebRtcIlbcfix_SimpleLsfQ(lsfdeq, lsf_index, lsf, iLBCenc_inst->lpc_n);
+
+  /* Stableize the LSF's if needed */
+  WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCenc_inst->lpc_n);
+
+  /* Calculate the synthesis and weighting filter coefficients from
+     the optimal LSF and the dequantized LSF */
+  WebRtcIlbcfix_SimpleInterpolateLsf(syntdenum, weightdenum,
+                                     lsf, lsfdeq, iLBCenc_inst->lsfold,
+                                     iLBCenc_inst->lsfdeqold, LPC_FILTERORDER, iLBCenc_inst);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
new file mode 100644
index 0000000000..a67b77acbf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LpcEncode.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc encoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LpcEncode(
+    int16_t* syntdenum,   /* (i/o) synthesis filter coefficients
+                                   before/after encoding */
+    int16_t* weightdenum, /* (i/o) weighting denumerator coefficients
+                                   before/after encoding */
+    int16_t* lsf_index,   /* (o) lsf quantization index */
+    int16_t* data,        /* (i) Speech to do LPC analysis on */
+    IlbcEncoder* iLBCenc_inst
+    /* (i/o) the encoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.c
new file mode 100644
index 0000000000..9f0e19a2d9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.c
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfCheck.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  check for stability of lsf coefficients
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_LsfCheck(
+    int16_t *lsf, /* LSF parameters */
+    int dim, /* dimension of LSF */
+    int NoAn)  /* No of analysis per frame */
+{
+  int k,n,m, Nit=2, change=0,pos;
+  const int16_t eps=319;  /* 0.039 in Q13 (50 Hz)*/
+  const int16_t eps2=160;  /* eps/2.0 in Q13;*/
+  const int16_t maxlsf=25723; /* 3.14; (4000 Hz)*/
+  const int16_t minlsf=82;  /* 0.01; (0 Hz)*/
+
+  /* LSF separation check*/
+  for (n=0;n<Nit;n++) {  /* Run through a 2 times */
+    for (m=0;m<NoAn;m++) { /* Number of analyses per frame */
+      for (k=0;k<(dim-1);k++) {
+        pos=m*dim+k;
+
+        /* Seperate coefficients with a safety margin of 50 Hz */
+        if ((lsf[pos+1]-lsf[pos])<eps) {
+
+          if (lsf[pos+1]<lsf[pos]) {
+            lsf[pos+1]= lsf[pos]+eps2;
+            lsf[pos]= lsf[pos+1]-eps2;
+          } else {
+            lsf[pos]-=eps2;
+            lsf[pos+1]+=eps2;
+          }
+          change=1;
+        }
+
+        /* Limit minimum and maximum LSF */
+        if (lsf[pos]<minlsf) {
+          lsf[pos]=minlsf;
+          change=1;
+        }
+
+        if (lsf[pos]>maxlsf) {
+          lsf[pos]=maxlsf;
+          change=1;
+        }
+      }
+    }
+  }
+
+  return change;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.h
new file mode 100644
index 0000000000..9ba90a31e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_check.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfCheck.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  check for stability of lsf coefficients
+ *---------------------------------------------------------------*/
+
+int WebRtcIlbcfix_LsfCheck(int16_t* lsf, /* LSF parameters */
+                           int dim,      /* dimension of LSF */
+                           int NoAn);    /* No of analysis per frame */
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
new file mode 100644
index 0000000000..04de5e7e6c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LspInterpolate2PolyDec.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
+
+/*----------------------------------------------------------------*
+ *  interpolation of lsf coefficients for the decoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LspInterpolate2PolyDec(
+    int16_t *a,   /* (o) lpc coefficients Q12 */
+    int16_t *lsf1,  /* (i) first set of lsf coefficients Q13 */
+    int16_t *lsf2,  /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
+                                   lsf1 and lsf2 Q14 */
+    int16_t length  /* (i) length of coefficient vectors */
+                                          ){
+  int16_t lsftmp[LPC_FILTERORDER];
+
+  /* interpolate LSF */
+  WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
+
+  /* Compute the filter coefficients from the LSF */
+  WebRtcIlbcfix_Lsf2Poly(a, lsftmp);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
new file mode 100644
index 0000000000..6cc9d9746d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LspInterpolate2PolyDec.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  interpolation of lsf coefficients for the decoder
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LspInterpolate2PolyDec(
+    int16_t* a,    /* (o) lpc coefficients Q12 */
+    int16_t* lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t* lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
+                                   lsf1 and lsf2 Q14 */
+    int16_t length /* (i) length of coefficient vectors */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
new file mode 100644
index 0000000000..618821216c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator and conversion from lsf to a coefficients
+ *  (subrutine to SimpleInterpolateLSF)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
+    int16_t *a,  /* (o) lpc coefficients Q12 */
+    int16_t *lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t *lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef, /* (i) weighting coefficient to use between
+                           lsf1 and lsf2 Q14 */
+    int16_t length /* (i) length of coefficient vectors */
+                                          ) {
+  /* Stack based */
+  int16_t lsftmp[LPC_FILTERORDER];
+
+  /* interpolate LSF */
+  WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
+
+  /* Compute the filter coefficients from the LSF */
+  WebRtcIlbcfix_Lsf2Poly(a, lsftmp);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
new file mode 100644
index 0000000000..b278a10f4b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_LsfInterpolate2PloyEnc.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator and conversion from lsf to a coefficients
+ *  (subrutine to SimpleInterpolateLSF)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
+    int16_t* a,    /* (o) lpc coefficients Q12 */
+    int16_t* lsf1, /* (i) first set of lsf coefficients Q13 */
+    int16_t* lsf2, /* (i) second set of lsf coefficients Q13 */
+    int16_t coef,  /* (i) weighting coefficient to use between
+                            lsf1 and lsf2 Q14 */
+    int16_t length /* (i) length of coefficient vectors */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
new file mode 100644
index 0000000000..ee8292f394
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Lsp.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_to_lsp.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  conversion from lsf to lsp coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Lsp(
+    int16_t *lsf, /* (i) lsf in Q13 values between 0 and pi */
+    int16_t *lsp, /* (o) lsp in Q15 values between -1 and 1 */
+    int16_t m  /* (i) number of coefficients */
+                           ) {
+  int16_t i, k;
+  int16_t diff; /* difference, which is used for the
+                           linear approximation (Q8) */
+  int16_t freq; /* normalized frequency in Q15 (0..1) */
+  int32_t tmpW32;
+
+  for(i=0; i<m; i++)
+  {
+    freq = (int16_t)((lsf[i] * 20861) >> 15);
+    /* 20861: 1.0/(2.0*PI) in Q17 */
+    /*
+       Upper 8 bits give the index k and
+       Lower 8 bits give the difference, which needs
+       to be approximated linearly
+    */
+    k = freq >> 8;
+    diff = (freq&0x00ff);
+
+    /* Guard against getting outside table */
+
+    if (k>63) {
+      k = 63;
+    }
+
+    /* Calculate linear approximation */
+    tmpW32 = WebRtcIlbcfix_kCosDerivative[k] * diff;
+    lsp[i] = WebRtcIlbcfix_kCos[k] + (int16_t)(tmpW32 >> 12);
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
new file mode 100644
index 0000000000..6bc6c44dbd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Lsp.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  conversion from lsf to lsp coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Lsp(
+    int16_t* lsf, /* (i) lsf in Q13 values between 0 and pi */
+    int16_t* lsp, /* (o) lsp in Q15 values between -1 and 1 */
+    int16_t m     /* (i) number of coefficients */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
new file mode 100644
index 0000000000..8ca91d82f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Poly.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/get_lsp_poly.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_to_lsp.h"
+
+void WebRtcIlbcfix_Lsf2Poly(
+    int16_t *a,     /* (o) predictor coefficients (order = 10) in Q12 */
+    int16_t *lsf    /* (i) line spectral frequencies in Q13 */
+                            ) {
+  int32_t f[2][6]; /* f[0][] and f[1][] corresponds to
+                            F1(z) and F2(z) respectivly */
+  int32_t *f1ptr, *f2ptr;
+  int16_t *a1ptr, *a2ptr;
+  int32_t tmpW32;
+  int16_t lsp[10];
+  int i;
+
+  /* Convert lsf to lsp */
+  WebRtcIlbcfix_Lsf2Lsp(lsf, lsp, LPC_FILTERORDER);
+
+  /* Get F1(z) and F2(z) from the lsp */
+  f1ptr=f[0];
+  f2ptr=f[1];
+  WebRtcIlbcfix_GetLspPoly(&lsp[0],f1ptr);
+  WebRtcIlbcfix_GetLspPoly(&lsp[1],f2ptr);
+
+  /* for i = 5 down to 1
+     Compute f1[i] += f1[i-1];
+     and     f2[i] += f2[i-1];
+  */
+  f1ptr=&f[0][5];
+  f2ptr=&f[1][5];
+  for (i=5; i>0; i--)
+  {
+    (*f1ptr) += (*(f1ptr-1));
+    (*f2ptr) -= (*(f2ptr-1));
+    f1ptr--;
+    f2ptr--;
+  }
+
+  /* Get the A(z) coefficients
+     a[0] = 1.0
+     for i = 1 to 5
+     a[i] = (f1[i] + f2[i] + round)>>13;
+     for i = 1 to 5
+     a[11-i] = (f1[i] - f2[i] + round)>>13;
+  */
+  a[0]=4096;
+  a1ptr=&a[1];
+  a2ptr=&a[10];
+  f1ptr=&f[0][1];
+  f2ptr=&f[1][1];
+  for (i=5; i>0; i--)
+  {
+    tmpW32 = (*f1ptr) + (*f2ptr);
+    *a1ptr = (int16_t)((tmpW32 + 4096) >> 13);
+
+    tmpW32 = (*f1ptr) - (*f2ptr);
+    *a2ptr = (int16_t)((tmpW32 + 4096) >> 13);
+
+    a1ptr++;
+    a2ptr--;
+    f1ptr++;
+    f2ptr++;
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
new file mode 100644
index 0000000000..f26d3a8d2d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsf2Poly.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Convert from LSF coefficients to A coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsf2Poly(
+    int16_t* a,  /* (o) predictor coefficients (order = 10) in Q12 */
+    int16_t* lsf /* (i) line spectral frequencies in Q13 */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
new file mode 100644
index 0000000000..227f4d45b4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsp2Lsf.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/lsp_to_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  conversion from LSP coefficients to LSF coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsp2Lsf(
+    int16_t *lsp, /* (i) lsp vector -1...+1 in Q15 */
+    int16_t *lsf, /* (o) Lsf vector 0...Pi in Q13
+                           (ordered, so that lsf[i]<lsf[i+1]) */
+    int16_t m  /* (i) Number of coefficients */
+                           )
+{
+  int16_t i, k;
+  int16_t diff; /* diff between table value and desired value (Q15) */
+  int16_t freq; /* lsf/(2*pi) (Q16) */
+  int16_t *lspPtr, *lsfPtr, *cosTblPtr;
+  int16_t tmp;
+
+  /* set the index to maximum index value in WebRtcIlbcfix_kCos */
+  k = 63;
+
+  /*
+     Start with the highest LSP and then work the way down
+     For each LSP the lsf is calculated by first order approximation
+     of the acos(x) function
+  */
+  lspPtr = &lsp[9];
+  lsfPtr = &lsf[9];
+  cosTblPtr=(int16_t*)&WebRtcIlbcfix_kCos[k];
+  for(i=m-1; i>=0; i--)
+  {
+    /*
+       locate value in the table, which is just above lsp[i],
+       basically an approximation to acos(x)
+    */
+    while( (((int32_t)(*cosTblPtr)-(*lspPtr)) < 0)&&(k>0) )
+    {
+      k-=1;
+      cosTblPtr--;
+    }
+
+    /* Calculate diff, which is used in the linear approximation of acos(x) */
+    diff = (*lspPtr)-(*cosTblPtr);
+
+    /*
+       The linear approximation of acos(lsp[i]) :
+       acos(lsp[i])= k*512 + (WebRtcIlbcfix_kAcosDerivative[ind]*offset >> 11)
+    */
+
+    /* tmp (linear offset) in Q16 */
+    tmp = (int16_t)((WebRtcIlbcfix_kAcosDerivative[k] * diff) >> 11);
+
+    /* freq in Q16 */
+    freq = (k << 9) + tmp;
+
+    /* lsf = freq*2*pi */
+    (*lsfPtr) = (int16_t)(((int32_t)freq*25736)>>15);
+
+    lsfPtr--;
+    lspPtr--;
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
new file mode 100644
index 0000000000..c2f4b7692d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Lsp2Lsf.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  conversion from LSP coefficients to LSF coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Lsp2Lsf(
+    int16_t* lsp, /* (i) lsp vector -1...+1 in Q15 */
+    int16_t* lsf, /* (o) Lsf vector 0...Pi in Q13
+                           (ordered, so that lsf[i]<lsf[i+1]) */
+    int16_t m     /* (i) Number of coefficients */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.c
new file mode 100644
index 0000000000..9b870e0ef0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.c
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_MyCorr.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/my_corr.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * compute cross correlation between sequences
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_MyCorr(
+    int32_t* corr,  /* (o) correlation of seq1 and seq2 */
+    const int16_t* seq1,  /* (i) first sequence */
+    size_t dim1,  /* (i) dimension first seq1 */
+    const int16_t* seq2, /* (i) second sequence */
+    size_t dim2   /* (i) dimension seq2 */
+                          ){
+  uint32_t max1, max2;
+  size_t loops;
+  int right_shift;
+
+  // Calculate a right shift that will let us sum dim2 pairwise products of
+  // values from the two sequences without overflowing an int32_t. (The +1 in
+  // max1 and max2 are because WebRtcSpl_MaxAbsValueW16 will return 2**15 - 1
+  // if the input array contains -2**15.)
+  max1 = WebRtcSpl_MaxAbsValueW16(seq1, dim1) + 1;
+  max2 = WebRtcSpl_MaxAbsValueW16(seq2, dim2) + 1;
+  right_shift =
+      (64 - 31) - WebRtcSpl_CountLeadingZeros64((max1 * max2) * (uint64_t)dim2);
+  if (right_shift < 0) {
+    right_shift = 0;
+  }
+
+  loops=dim1-dim2+1;
+
+  /* Calculate the cross correlations */
+  WebRtcSpl_CrossCorrelation(corr, seq2, seq1, dim2, loops, right_shift, 1);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.h
new file mode 100644
index 0000000000..c0c2fa4a48
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/my_corr.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_MyCorr.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * compute cross correlation between sequences
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_MyCorr(int32_t* corr, /* (o) correlation of seq1 and seq2 */
+                          const int16_t* seq1, /* (i) first sequence */
+                          size_t dim1,         /* (i) dimension first seq1 */
+                          const int16_t* seq2, /* (i) second sequence */
+                          size_t dim2          /* (i) dimension seq2 */
+                          );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
new file mode 100644
index 0000000000..1ecdd96d5a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_NearestNeighbor.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/nearest_neighbor.h"
+
+void WebRtcIlbcfix_NearestNeighbor(size_t* index,
+                                   const size_t* array,
+                                   size_t value,
+                                   size_t arlength) {
+  size_t i;
+  size_t min_diff = (size_t)-1;
+  for (i = 0; i < arlength; i++) {
+    const size_t diff =
+        (array[i] < value) ? (value - array[i]) : (array[i] - value);
+    if (diff < min_diff) {
+      *index = i;
+      min_diff = diff;
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
new file mode 100644
index 0000000000..704cf2a37d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_NearestNeighbor.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * Find index in array such that the array element with said
+ * index is the element of said array closest to "value"
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_NearestNeighbor(
+    size_t* index,       /* (o) index of array element closest to value */
+    const size_t* array, /* (i) data array (Q2) */
+    size_t value,        /* (i) value (Q2) */
+    size_t arlength      /* (i) dimension of data array (==ENH_NBLOCKS_TOT) */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.c
new file mode 100644
index 0000000000..dd44eb8fb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.c
@@ -0,0 +1,253 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_PackBits.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/pack_bits.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_PackBits(
+    uint16_t *bitstream,   /* (o) The packetized bitstream */
+    iLBC_bits *enc_bits,  /* (i) Encoded bits */
+    int16_t mode     /* (i) Codec mode (20 or 30) */
+                             ){
+  uint16_t *bitstreamPtr;
+  int i, k;
+  int16_t *tmpPtr;
+
+  bitstreamPtr=bitstream;
+
+  /* Class 1 bits of ULP */
+  /* First int16_t */
+  (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[0])<<10;   /* Bit 0..5  */
+  (*bitstreamPtr) |= (enc_bits->lsf[1])<<3;     /* Bit 6..12 */
+  (*bitstreamPtr) |= (enc_bits->lsf[2]&0x70)>>4;    /* Bit 13..15 */
+  bitstreamPtr++;
+  /* Second int16_t */
+  (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[2]&0xF)<<12;  /* Bit 0..3  */
+
+  if (mode==20) {
+    (*bitstreamPtr) |= (enc_bits->startIdx)<<10;    /* Bit 4..5  */
+    (*bitstreamPtr) |= (enc_bits->state_first)<<9;    /* Bit 6  */
+    (*bitstreamPtr) |= (enc_bits->idxForMax)<<3;    /* Bit 7..12 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[0])&0x70)>>4;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    (*bitstreamPtr) = ((enc_bits->cb_index[0])&0xE)<<12;  /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x18)<<8;  /* Bit 3..4  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x8)<<7;  /* Bit 5  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0xFE)<<2;  /* Bit 6..12 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x10)>>2;  /* Bit 13  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x8)>>2;  /* Bit 14  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x10)>>4;  /* Bit 15  */
+  } else { /* mode==30 */
+    (*bitstreamPtr) |= (enc_bits->lsf[3])<<6;     /* Bit 4..9  */
+    (*bitstreamPtr) |= (enc_bits->lsf[4]&0x7E)>>1;    /* Bit 10..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->lsf[4]&0x1)<<15;  /* Bit 0  */
+    (*bitstreamPtr) |= (enc_bits->lsf[5])<<8;     /* Bit 1..7  */
+    (*bitstreamPtr) |= (enc_bits->startIdx)<<5;     /* Bit 8..10 */
+    (*bitstreamPtr) |= (enc_bits->state_first)<<4;    /* Bit 11  */
+    (*bitstreamPtr) |= ((enc_bits->idxForMax)&0x3C)>>2;   /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 4:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->idxForMax&0x3)<<14; /* Bit 0..1  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x78)<<7;   /* Bit 2..5  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x10)<<5;  /* Bit 6  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x8)<<5;  /* Bit 7  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[3]&0xFC);   /* Bit 8..13 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[3]&0x10)>>3;  /* Bit 14  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x8)>>3;  /* Bit 15  */
+  }
+  /* Class 2 bits of ULP */
+  /* 4:th to 6:th int16_t for 20 ms case
+     5:th to 7:th int16_t for 30 ms case */
+  bitstreamPtr++;
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<3; k++) {
+    (*bitstreamPtr) = 0;
+    for (i=15; i>=0; i--) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 7:th int16_t */
+    (*bitstreamPtr) = 0;
+    for (i=15; i>6; i--) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4)<<4;  /* Bit 9  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<2;  /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x4)<<1;  /* Bit 12  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x8)>>1;  /* Bit 13  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)>>2;  /* Bit 14..15 */
+
+  } else { /* mode==30 */
+    /* 8:th int16_t */
+    (*bitstreamPtr) = 0;
+    for (i=15; i>5; i--) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x4)>>2)<<i;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x6)<<3;   /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x8);   /* Bit 12  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4);   /* Bit 13  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[3]&0x2);    /* Bit 14  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[6]&0x80)>>7;   /* Bit 15  */
+    bitstreamPtr++;
+    /* 9:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->cb_index[6]&0x7E)<<9;/* Bit 0..5  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[9]&0xFE)<<2;   /* Bit 6..12 */
+    (*bitstreamPtr) |= (enc_bits->cb_index[12]&0xE0)>>5;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 10:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)enc_bits->cb_index[12]&0x1E)<<11;/* Bit 0..3 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<8;  /* Bit 4..5  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x6)<<7;  /* Bit 6..7  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x18)<<3;  /* Bit 8..9  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)<<2;  /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[9]&0x10)>>1;  /* Bit 12  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[10]&0x8)>>1;  /* Bit 13  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[12]&0x10)>>3;  /* Bit 14  */
+    (*bitstreamPtr) |= (enc_bits->gain_index[13]&0x8)>>3;  /* Bit 15  */
+  }
+  bitstreamPtr++;
+  /* Class 3 bits of ULP */
+  /*  8:th to 14:th int16_t for 20 ms case
+      11:th to 17:th int16_t for 30 ms case */
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<7; k++) {
+    (*bitstreamPtr) = 0;
+    for (i=14; i>=0; i-=2) {
+      (*bitstreamPtr) |= ((uint16_t)((*tmpPtr)&0x3))<<i; /* Bit 15-i..14-i*/
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 15:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
+    (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<13;  /* Bit 2  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<6;   /* Bit 3..9  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x7E)>>1;  /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 16:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->cb_index[2])&0x1))<<15;
+    /* Bit 0  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<12;  /* Bit 1..3  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<10;  /* Bit 4..5  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[2]))<<7;   /* Bit 6..8  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<6;  /* Bit 9  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x7E)>>1;  /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 17:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->cb_index[4])&0x1))<<15;
+    /* Bit 0  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[5])<<8;    /* Bit 1..7  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[6]);     /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 18:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)(enc_bits->cb_index[7]))<<8; /* Bit 0..7  */
+    (*bitstreamPtr) |= (enc_bits->cb_index[8]);     /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    (*bitstreamPtr) = ((uint16_t)((enc_bits->gain_index[3])&0x3))<<14;
+    /* Bit 0..1  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x3)<<12;  /* Bit 2..3  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[5]))<<9;   /* Bit 4..6  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<6;  /* Bit 7..9  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<4;  /* Bit 10..11 */
+    (*bitstreamPtr) |= (enc_bits->gain_index[8])<<1;   /* Bit 12..14 */
+  } else { /* mode==30 */
+    /* 18:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
+    (*bitstreamPtr) |= (((enc_bits->idxVec[57])&0x3))<<12;  /* Bit 2..3  */
+    (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<11;  /* Bit 4  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<4;   /* Bit 5..11 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x78)>>3;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[2])&0x7)<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<10;  /* Bit 3..5  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<8;  /* Bit 6..7  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[2])&0x7)<<5;  /* Bit 8..10 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<4;  /* Bit 11  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x78)>>3;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 20:th int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[4])&0x7)<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[5]))<<6;   /* Bit 3..9  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[6])&0x1)<<5;  /* Bit 10  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[7])&0xF8)>>3;  /* Bit 11..15 */
+    bitstreamPtr++;
+    /* 21:st int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[7])&0x7)<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[8]))<<5;   /* Bit 3..10 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[9])&0x1)<<4;  /* Bit 11  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[10])&0xF0)>>4;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 22:nd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[10])&0xF)<<12;
+    /* Bit 0..3  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[11]))<<4;   /* Bit 4..11 */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[12])&0x1)<<3;  /* Bit 12  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[13])&0xE0)>>5;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 23:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->cb_index[13])&0x1F)<<11;
+    /* Bit 0..4  */
+    (*bitstreamPtr) |= ((enc_bits->cb_index[14]))<<3;   /* Bit 5..12 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x3)<<1;  /* Bit 13..14 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x1);   /* Bit 15  */
+    bitstreamPtr++;
+    /* 24:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->gain_index[5]))<<13;
+    /* Bit 0..2  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<10;  /* Bit 3..5  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<8;  /* Bit 6..7  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[8]))<<5;   /* Bit 8..10 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[9])&0xF)<<1;  /* Bit 11..14 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[10])&0x4)>>2;  /* Bit 15  */
+    bitstreamPtr++;
+    /* 25:rd int16_t */
+    (*bitstreamPtr)  = ((uint16_t)(enc_bits->gain_index[10])&0x3)<<14;
+    /* Bit 0..1  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[11]))<<11;  /* Bit 2..4  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[12])&0xF)<<7;  /* Bit 5..8  */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[13])&0x7)<<4;  /* Bit 9..11 */
+    (*bitstreamPtr) |= ((enc_bits->gain_index[14]))<<1;   /* Bit 12..14 */
+  }
+  /* Last bit is automatically zero */
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.h
new file mode 100644
index 0000000000..8dcf41ce08
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/pack_bits.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_PackBits.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_PackBits(
+    uint16_t* bitstream, /* (o) The packetized bitstream */
+    iLBC_bits* enc_bits, /* (i) Encoded bits */
+    int16_t mode         /* (i) Codec mode (20 or 30) */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
new file mode 100644
index 0000000000..7192eaab49
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsf.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/lsp_to_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsp.h"
+
+void WebRtcIlbcfix_Poly2Lsf(
+    int16_t *lsf,   /* (o) lsf coefficients (Q13) */
+    int16_t *a    /* (i) A coefficients (Q12) */
+                            ) {
+  int16_t lsp[10];
+  WebRtcIlbcfix_Poly2Lsp(a, lsp, (int16_t*)WebRtcIlbcfix_kLspMean);
+  WebRtcIlbcfix_Lsp2Lsf(lsp, lsf, 10);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
new file mode 100644
index 0000000000..363e392bb2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsf.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  conversion from lpc coefficients to lsf coefficients
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsf(int16_t* lsf, /* (o) lsf coefficients (Q13) */
+                            int16_t* a    /* (i) A coefficients (Q12) */
+                            );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
new file mode 100644
index 0000000000..ad0ecd70ab
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
@@ -0,0 +1,159 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsp.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsp.h"
+
+#include "modules/audio_coding/codecs/ilbc/chebyshev.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+/*----------------------------------------------------------------*
+ * conversion from lpc coefficients to lsp coefficients
+ * function is only for 10:th order LPC
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsp(
+    int16_t *a,  /* (o) A coefficients in Q12 */
+    int16_t *lsp, /* (i) LSP coefficients in Q15 */
+    int16_t *old_lsp /* (i) old LSP coefficients that are used if the new
+                              coefficients turn out to be unstable */
+                            ) {
+  int16_t f[2][6]; /* f[0][] represents f1 and f[1][] represents f2 */
+  int16_t *a_i_ptr, *a_10mi_ptr;
+  int16_t *f1ptr, *f2ptr;
+  int32_t tmpW32;
+  int16_t x, y, xlow, ylow, xmid, ymid, xhigh, yhigh, xint;
+  int16_t shifts, sign;
+  int i, j;
+  int foundFreqs;
+  int fi_select;
+
+  /*
+     Calculate the two polynomials f1(z) and f2(z)
+     (the sum and the diff polynomial)
+     f1[0] = f2[0] = 1.0;
+     f1[i+1] = a[i+1] + a[10-i] - f1[i];
+     f2[i+1] = a[i+1] - a[10-i] - f1[i];
+  */
+
+  a_i_ptr = a + 1;
+  a_10mi_ptr = a + 10;
+  f1ptr = f[0];
+  f2ptr = f[1];
+  (*f1ptr) = 1024; /* 1.0 in Q10 */
+  (*f2ptr) = 1024; /* 1.0 in Q10 */
+  for (i = 0; i < 5; i++) {
+    *(f1ptr + 1) =
+        (int16_t)((((int32_t)(*a_i_ptr) + *a_10mi_ptr) >> 2) - *f1ptr);
+    *(f2ptr + 1) =
+        (int16_t)((((int32_t)(*a_i_ptr) - *a_10mi_ptr) >> 2) + *f2ptr);
+    a_i_ptr++;
+    a_10mi_ptr--;
+    f1ptr++;
+    f2ptr++;
+  }
+
+  /*
+    find the LSPs using the Chebychev pol. evaluation
+  */
+
+  fi_select = 0; /* selector between f1 and f2, start with f1 */
+
+  foundFreqs = 0;
+
+  xlow = WebRtcIlbcfix_kCosGrid[0];
+  ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+
+  /*
+     Iterate until all the 10 LSP's have been found or
+     all the grid points have been tried. If the 10 LSP's can
+     not be found, set the LSP vector to previous LSP
+  */
+
+  for (j = 1; j < COS_GRID_POINTS && foundFreqs < 10; j++) {
+    xhigh = xlow;
+    yhigh = ylow;
+    xlow = WebRtcIlbcfix_kCosGrid[j];
+    ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+
+    if (ylow * yhigh <= 0) {
+      /* Run 4 times to reduce the interval */
+      for (i = 0; i < 4; i++) {
+        /* xmid =(xlow + xhigh)/2 */
+        xmid = (xlow >> 1) + (xhigh >> 1);
+        ymid = WebRtcIlbcfix_Chebyshev(xmid, f[fi_select]);
+
+        if (ylow * ymid <= 0) {
+          yhigh = ymid;
+          xhigh = xmid;
+        } else {
+          ylow = ymid;
+          xlow = xmid;
+        }
+      }
+
+      /*
+        Calculater xint by linear interpolation:
+        xint = xlow - ylow*(xhigh-xlow)/(yhigh-ylow);
+      */
+
+      x = xhigh - xlow;
+      y = yhigh - ylow;
+
+      if (y == 0) {
+        xint = xlow;
+      } else {
+        sign = y;
+        y = WEBRTC_SPL_ABS_W16(y);
+        shifts = (int16_t)WebRtcSpl_NormW32(y)-16;
+        y <<= shifts;
+        y = (int16_t)WebRtcSpl_DivW32W16(536838144, y); /* 1/(yhigh-ylow) */
+
+        tmpW32 = (x * y) >> (19 - shifts);
+
+        /* y=(xhigh-xlow)/(yhigh-ylow) */
+        y = (int16_t)(tmpW32&0xFFFF);
+
+        if (sign < 0) {
+          y = -y;
+        }
+        /* tmpW32 = ylow*(xhigh-xlow)/(yhigh-ylow) */
+        tmpW32 = (ylow * y) >> 10;
+        xint = xlow-(int16_t)(tmpW32&0xFFFF);
+      }
+
+      /* Store the calculated lsp */
+      lsp[foundFreqs] = (int16_t)xint;
+      foundFreqs++;
+
+      /* if needed, set xlow and ylow for next recursion */
+      if (foundFreqs<10) {
+        xlow = xint;
+        /* Swap between f1 and f2 (f[0][] and f[1][]) */
+        fi_select = ((fi_select+1)&0x1);
+
+        ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
+      }
+    }
+  }
+
+  /* Check if M roots found, if not then use the old LSP */
+  if (foundFreqs < 10) {
+    WEBRTC_SPL_MEMCPY_W16(lsp, old_lsp, 10);
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
new file mode 100644
index 0000000000..928ee4efdb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Poly2Lsp.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * conversion from lpc coefficients to lsp coefficients
+ * function is only for 10:th order LPC
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Poly2Lsp(
+    int16_t* a,      /* (o) A coefficients in Q12 */
+    int16_t* lsp,    /* (i) LSP coefficients in Q15 */
+    int16_t* old_lsp /* (i) old LSP coefficients that are used if the new
+                              coefficients turn out to be unstable */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.c
new file mode 100644
index 0000000000..5bdab7a4b0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.c
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Refiner.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/refiner.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/enh_upsample.h"
+#include "modules/audio_coding/codecs/ilbc/my_corr.h"
+
+/*----------------------------------------------------------------*
+ * find segment starting near idata+estSegPos that has highest
+ * correlation with idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
+ * resolution of ENH_UPSO times the original of the original
+ * sampling rate
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Refiner(
+    size_t *updStartPos, /* (o) updated start point (Q-2) */
+    int16_t *idata,   /* (i) original data buffer */
+    size_t idatal,   /* (i) dimension of idata */
+    size_t centerStartPos, /* (i) beginning center segment */
+    size_t estSegPos,  /* (i) estimated beginning other segment (Q-2) */
+    int16_t *surround,  /* (i/o) The contribution from this sequence
+                                           summed with earlier contributions */
+    int16_t gain    /* (i) Gain to use for this sequence */
+                           ){
+  size_t estSegPosRounded, searchSegStartPos, searchSegEndPos, corrdim;
+  size_t tloc, tloc2, i;
+
+  int32_t maxtemp, scalefact;
+  int16_t *filtStatePtr, *polyPtr;
+  /* Stack based */
+  int16_t filt[7];
+  int32_t corrVecUps[ENH_CORRDIM*ENH_UPS0];
+  int32_t corrVecTemp[ENH_CORRDIM];
+  int16_t vect[ENH_VECTL];
+  int16_t corrVec[ENH_CORRDIM];
+
+  /* defining array bounds */
+
+  estSegPosRounded = (estSegPos - 2) >> 2;
+
+  searchSegStartPos =
+      (estSegPosRounded < ENH_SLOP) ? 0 : (estSegPosRounded - ENH_SLOP);
+
+  searchSegEndPos = estSegPosRounded + ENH_SLOP;
+  if ((searchSegEndPos + ENH_BLOCKL) >= idatal) {
+    searchSegEndPos = idatal - ENH_BLOCKL - 1;
+  }
+
+  corrdim = searchSegEndPos + 1 - searchSegStartPos;
+
+  /* compute upsampled correlation and find
+     location of max */
+
+  WebRtcIlbcfix_MyCorr(corrVecTemp, idata + searchSegStartPos,
+                       corrdim + ENH_BLOCKL - 1, idata + centerStartPos,
+                       ENH_BLOCKL);
+
+  /* Calculate the rescaling factor for the correlation in order to
+     put the correlation in a int16_t vector instead */
+  maxtemp = WebRtcSpl_MaxAbsValueW32(corrVecTemp, corrdim);
+
+  scalefact = WebRtcSpl_GetSizeInBits(maxtemp) - 15;
+
+  if (scalefact > 0) {
+    for (i = 0; i < corrdim; i++) {
+      corrVec[i] = (int16_t)(corrVecTemp[i] >> scalefact);
+    }
+  } else {
+    for (i = 0; i < corrdim; i++) {
+      corrVec[i] = (int16_t)corrVecTemp[i];
+    }
+  }
+  /* In order to guarantee that all values are initialized */
+  for (i = corrdim; i < ENH_CORRDIM; i++) {
+    corrVec[i] = 0;
+  }
+
+  /* Upsample the correlation */
+  WebRtcIlbcfix_EnhUpsample(corrVecUps, corrVec);
+
+  /* Find maximum */
+  tloc = WebRtcSpl_MaxIndexW32(corrVecUps, ENH_UPS0 * corrdim);
+
+  /* make vector can be upsampled without ever running outside
+     bounds */
+  *updStartPos = searchSegStartPos * 4 + tloc + 4;
+
+  tloc2 = (tloc + 3) >> 2;
+
+  /* initialize the vector to be filtered, stuff with zeros
+     when data is outside idata buffer */
+  if (ENH_FL0 > (searchSegStartPos + tloc2)) {
+    const size_t st = ENH_FL0 - searchSegStartPos - tloc2;
+    WebRtcSpl_MemSetW16(vect, 0, st);
+    WEBRTC_SPL_MEMCPY_W16(&vect[st], idata, ENH_VECTL - st);
+  } else {
+    const size_t st = searchSegStartPos + tloc2 - ENH_FL0;
+    if ((st + ENH_VECTL) > idatal) {
+      const size_t en = st + ENH_VECTL - idatal;
+      WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL - en);
+      WebRtcSpl_MemSetW16(&vect[ENH_VECTL - en], 0, en);
+    } else {
+      WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL);
+    }
+  }
+
+  /* compute the segment (this is actually a convolution) */
+  filtStatePtr = filt + 6;
+  polyPtr = (int16_t*)WebRtcIlbcfix_kEnhPolyPhaser[tloc2 * ENH_UPS0 - tloc];
+  for (i = 0; i < 7; i++) {
+    *filtStatePtr-- = *polyPtr++;
+  }
+
+  WebRtcSpl_FilterMAFastQ12(&vect[6], vect, filt, ENH_FLO_MULT2_PLUS1,
+                            ENH_BLOCKL);
+
+  /* Add the contribution from this vector (scaled with gain) to the total
+     surround vector */
+  WebRtcSpl_AddAffineVectorToVector(surround, vect, gain, 32768, 16,
+                                    ENH_BLOCKL);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.h
new file mode 100644
index 0000000000..564c9d96e6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/refiner.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Refiner.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * find segment starting near idata+estSegPos that has highest
+ * correlation with idata+centerStartPos through
+ * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
+ * resolution of ENH_UPSO times the original of the original
+ * sampling rate
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Refiner(
+    size_t* updStartPos,   /* (o) updated start point (Q-2) */
+    int16_t* idata,        /* (i) original data buffer */
+    size_t idatal,         /* (i) dimension of idata */
+    size_t centerStartPos, /* (i) beginning center segment */
+    size_t estSegPos,      /* (i) estimated beginning other segment (Q-2) */
+    int16_t* surround,     /* (i/o) The contribution from this sequence
+                                    summed with earlier contributions */
+    int16_t gain           /* (i) Gain to use for this sequence */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
new file mode 100644
index 0000000000..7343530a5e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleInterpolateLsf.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h"
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleInterpolateLsf(
+    int16_t *syntdenum, /* (o) the synthesis filter denominator
+                                   resulting from the quantized
+                                   interpolated lsf Q12 */
+    int16_t *weightdenum, /* (o) the weighting filter denominator
+                                   resulting from the unquantized
+                                   interpolated lsf Q12 */
+    int16_t *lsf,  /* (i) the unquantized lsf coefficients Q13 */
+    int16_t *lsfdeq,  /* (i) the dequantized lsf coefficients Q13 */
+    int16_t *lsfold,  /* (i) the unquantized lsf coefficients of
+                                           the previous signal frame Q13 */
+    int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the
+                                   previous signal frame Q13 */
+    int16_t length,  /* (i) should equate FILTERORDER */
+    IlbcEncoder *iLBCenc_inst
+    /* (i/o) the encoder state structure */
+                                        ) {
+  size_t i;
+  int pos, lp_length;
+
+  int16_t *lsf2, *lsfdeq2;
+  /* Stack based */
+  int16_t lp[LPC_FILTERORDER + 1];
+
+  lsf2 = lsf + length;
+  lsfdeq2 = lsfdeq + length;
+  lp_length = length + 1;
+
+  if (iLBCenc_inst->mode==30) {
+    /* subframe 1: Interpolation between old and first set of
+       lsf coefficients */
+
+    /* Calculate Analysis/Syntehsis filter from quantized LSF */
+    WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq,
+                                         WebRtcIlbcfix_kLsfWeight30ms[0],
+                                         length);
+    WEBRTC_SPL_MEMCPY_W16(syntdenum, lp, lp_length);
+
+    /* Calculate Weighting filter from quantized LSF */
+    WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf,
+                                         WebRtcIlbcfix_kLsfWeight30ms[0],
+                                         length);
+    WebRtcIlbcfix_BwExpand(weightdenum, lp,
+                           (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                           (int16_t)lp_length);
+
+    /* subframe 2 to 6: Interpolation between first and second
+       set of lsf coefficients */
+
+    pos = lp_length;
+    for (i = 1; i < iLBCenc_inst->nsub; i++) {
+
+      /* Calculate Analysis/Syntehsis filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeq, lsfdeq2,
+                                           WebRtcIlbcfix_kLsfWeight30ms[i],
+                                           length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length);
+
+      /* Calculate Weighting filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsf, lsf2,
+                                           WebRtcIlbcfix_kLsfWeight30ms[i],
+                                           length);
+      WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                             (int16_t)lp_length);
+
+      pos += lp_length;
+    }
+
+    /* update memory */
+
+    WEBRTC_SPL_MEMCPY_W16(lsfold, lsf2, length);
+    WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq2, length);
+
+  } else { /* iLBCenc_inst->mode==20 */
+    pos = 0;
+    for (i = 0; i < iLBCenc_inst->nsub; i++) {
+
+      /* Calculate Analysis/Syntehsis filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq,
+                                           WebRtcIlbcfix_kLsfWeight20ms[i],
+                                           length);
+      WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length);
+
+      /* Calculate Weighting filter from quantized LSF */
+      WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf,
+                                           WebRtcIlbcfix_kLsfWeight20ms[i],
+                                           length);
+      WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
+                             (int16_t*)WebRtcIlbcfix_kLpcChirpWeightDenum,
+                             (int16_t)lp_length);
+
+      pos += lp_length;
+    }
+
+    /* update memory */
+
+    WEBRTC_SPL_MEMCPY_W16(lsfold, lsf, length);
+    WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq, length);
+
+  }
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
new file mode 100644
index 0000000000..ee53e4bd08
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleInterpolateLsf.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lsf interpolator (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleInterpolateLsf(
+    int16_t* syntdenum,   /* (o) the synthesis filter denominator
+                                     resulting from the quantized
+                                     interpolated lsf Q12 */
+    int16_t* weightdenum, /* (o) the weighting filter denominator
+                                   resulting from the unquantized
+                                   interpolated lsf Q12 */
+    int16_t* lsf,         /* (i) the unquantized lsf coefficients Q13 */
+    int16_t* lsfdeq,      /* (i) the dequantized lsf coefficients Q13 */
+    int16_t* lsfold,      /* (i) the unquantized lsf coefficients of
+                                               the previous signal frame Q13 */
+    int16_t* lsfdeqold,   /* (i) the dequantized lsf coefficients of the
+                                     previous signal frame Q13 */
+    int16_t length,       /* (i) should equate FILTERORDER */
+    IlbcEncoder* iLBCenc_inst
+    /* (i/o) the encoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
new file mode 100644
index 0000000000..fdc4553d95
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLpcAnalysis.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h"
+
+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/poly_to_lsf.h"
+#include "modules/audio_coding/codecs/ilbc/window32_w32.h"
+
+/*----------------------------------------------------------------*
+ *  lpc analysis (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLpcAnalysis(
+    int16_t *lsf,   /* (o) lsf coefficients */
+    int16_t *data,   /* (i) new block of speech */
+    IlbcEncoder *iLBCenc_inst
+    /* (i/o) the encoder state structure */
+                                     ) {
+  int k;
+  int scale;
+  size_t is;
+  int16_t stability;
+  /* Stack based */
+  int16_t A[LPC_FILTERORDER + 1];
+  int32_t R[LPC_FILTERORDER + 1];
+  int16_t windowedData[BLOCKL_MAX];
+  int16_t rc[LPC_FILTERORDER];
+
+  is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer+is,data,iLBCenc_inst->blockl);
+
+  /* No lookahead, last window is asymmetric */
+
+  for (k = 0; k < iLBCenc_inst->lpc_n; k++) {
+
+    is = LPC_LOOKBACK;
+
+    if (k < (iLBCenc_inst->lpc_n - 1)) {
+
+      /* Hanning table WebRtcIlbcfix_kLpcWin[] is in Q15-domain so the output is right-shifted 15 */
+      WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer, WebRtcIlbcfix_kLpcWin, BLOCKL_MAX, 15);
+    } else {
+
+      /* Hanning table WebRtcIlbcfix_kLpcAsymWin[] is in Q15-domain so the output is right-shifted 15 */
+      WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer+is, WebRtcIlbcfix_kLpcAsymWin, BLOCKL_MAX, 15);
+    }
+
+    /* Compute autocorrelation */
+    WebRtcSpl_AutoCorrelation(windowedData, BLOCKL_MAX, LPC_FILTERORDER, R, &scale);
+
+    /* Window autocorrelation vector */
+    WebRtcIlbcfix_Window32W32(R, R, WebRtcIlbcfix_kLpcLagWin, LPC_FILTERORDER + 1 );
+
+    /* Calculate the A coefficients from the Autocorrelation using Levinson Durbin algorithm */
+    stability=WebRtcSpl_LevinsonDurbin(R, A, rc, LPC_FILTERORDER);
+
+    /*
+       Set the filter to {1.0, 0.0, 0.0,...} if filter from Levinson Durbin algorithm is unstable
+       This should basically never happen...
+    */
+    if (stability!=1) {
+      A[0]=4096;
+      WebRtcSpl_MemSetW16(&A[1], 0, LPC_FILTERORDER);
+    }
+
+    /* Bandwidth expand the filter coefficients */
+    WebRtcIlbcfix_BwExpand(A, A, (int16_t*)WebRtcIlbcfix_kLpcChirpSyntDenum, LPC_FILTERORDER+1);
+
+    /* Convert from A to LSF representation */
+    WebRtcIlbcfix_Poly2Lsf(lsf + k*LPC_FILTERORDER, A);
+  }
+
+  is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
+  WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer,
+                        iLBCenc_inst->lpc_buffer+LPC_LOOKBACK+BLOCKL_MAX-is, is);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
new file mode 100644
index 0000000000..b5c839ba2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLpcAnalysis.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  lpc analysis (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLpcAnalysis(
+    int16_t* lsf,  /* (o) lsf coefficients */
+    int16_t* data, /* (i) new block of speech */
+    IlbcEncoder* iLBCenc_inst
+    /* (i/o) the encoder state structure */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
new file mode 100644
index 0000000000..e7494ceb59
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfDeQ.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  obtain dequantized lsf coefficients from quantization index
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfDeQ(
+    int16_t *lsfdeq,  /* (o) dequantized lsf coefficients */
+    int16_t *index,  /* (i) quantization index */
+    int16_t lpc_n  /* (i) number of LPCs */
+                                ){
+  int i, j, pos, cb_pos;
+
+  /* decode first LSF */
+
+  pos = 0;
+  cb_pos = 0;
+  for (i = 0; i < LSF_NSPLIT; i++) {
+    for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) {
+      lsfdeq[pos + j] = WebRtcIlbcfix_kLsfCb[cb_pos + j + index[i] *
+                                             WebRtcIlbcfix_kLsfDimCb[i]];
+    }
+    pos += WebRtcIlbcfix_kLsfDimCb[i];
+    cb_pos += WebRtcIlbcfix_kLsfSizeCb[i] * WebRtcIlbcfix_kLsfDimCb[i];
+  }
+
+  if (lpc_n>1) {
+    /* decode last LSF */
+    pos = 0;
+    cb_pos = 0;
+    for (i = 0; i < LSF_NSPLIT; i++) {
+      for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) {
+        lsfdeq[LPC_FILTERORDER + pos + j] = WebRtcIlbcfix_kLsfCb[
+            cb_pos + index[LSF_NSPLIT + i] * WebRtcIlbcfix_kLsfDimCb[i] + j];
+      }
+      pos += WebRtcIlbcfix_kLsfDimCb[i];
+      cb_pos += WebRtcIlbcfix_kLsfSizeCb[i] * WebRtcIlbcfix_kLsfDimCb[i];
+    }
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
new file mode 100644
index 0000000000..6d97d3df33
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfDeQ.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  obtain dequantized lsf coefficients from quantization index
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfDeQ(
+    int16_t* lsfdeq, /* (o) dequantized lsf coefficients */
+    int16_t* index,  /* (i) quantization index */
+    int16_t lpc_n    /* (i) number of LPCs */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
new file mode 100644
index 0000000000..1291d1442e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfQ.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/simple_lsf_quant.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/split_vq.h"
+
+/*----------------------------------------------------------------*
+ *  lsf quantizer (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfQ(
+    int16_t *lsfdeq, /* (o) dequantized lsf coefficients
+                                   (dimension FILTERORDER) Q13 */
+    int16_t *index, /* (o) quantization index */
+    int16_t *lsf, /* (i) the lsf coefficient vector to be
+                           quantized (dimension FILTERORDER) Q13 */
+    int16_t lpc_n /* (i) number of lsf sets to quantize */
+                              ){
+
+  /* Quantize first LSF with memoryless split VQ */
+  WebRtcIlbcfix_SplitVq( lsfdeq, index, lsf,
+                         (int16_t*)WebRtcIlbcfix_kLsfCb, (int16_t*)WebRtcIlbcfix_kLsfDimCb, (int16_t*)WebRtcIlbcfix_kLsfSizeCb);
+
+  if (lpc_n==2) {
+    /* Quantize second LSF with memoryless split VQ */
+    WebRtcIlbcfix_SplitVq( lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT,
+                           lsf + LPC_FILTERORDER, (int16_t*)WebRtcIlbcfix_kLsfCb,
+                           (int16_t*)WebRtcIlbcfix_kLsfDimCb, (int16_t*)WebRtcIlbcfix_kLsfSizeCb);
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
new file mode 100644
index 0000000000..66b553213a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SimpleLsfQ.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  lsf quantizer (subrutine to LPCencode)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SimpleLsfQ(
+    int16_t* lsfdeq, /* (o) dequantized lsf coefficients
+                                   (dimension FILTERORDER) Q13 */
+    int16_t* index,  /* (o) quantization index */
+    int16_t* lsf,    /* (i) the lsf coefficient vector to be
+                              quantized (dimension FILTERORDER) Q13 */
+    int16_t lpc_n    /* (i) number of lsf sets to quantize */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.c
new file mode 100644
index 0000000000..631b2f432a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.c
@@ -0,0 +1,212 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/smooth.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/smooth_out_data.h"
+
+/*----------------------------------------------------------------*
+ * find the smoothed output data
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Smooth(
+    int16_t *odata,   /* (o) smoothed output */
+    int16_t *current,  /* (i) the un enhanced residual for
+                                this block */
+    int16_t *surround  /* (i) The approximation from the
+                                surrounding sequences */
+                          ) {
+  int16_t scale, scale1, scale2;
+  int16_t A, B, C, denomW16;
+  int32_t B_W32, denom, num;
+  int32_t errs;
+  int32_t w00,w10,w11, endiff, crit;
+  int32_t w00prim, w10prim, w11_div_w00;
+  int16_t w11prim;
+  int16_t bitsw00, bitsw10, bitsw11;
+  int32_t w11w00, w10w10, w00w00;
+  uint32_t max1, max2, max12;
+
+  /* compute some inner products (ensure no overflow by first calculating proper scale factor) */
+
+  w00 = w10 = w11 = 0;
+
+  // Calculate a right shift that will let us sum ENH_BLOCKL pairwise products
+  // of values from the two sequences without overflowing an int32_t. (The +1
+  // in max1 and max2 are because WebRtcSpl_MaxAbsValueW16 will return 2**15 -
+  // 1 if the input array contains -2**15.)
+  max1 = WebRtcSpl_MaxAbsValueW16(current, ENH_BLOCKL) + 1;
+  max2 = WebRtcSpl_MaxAbsValueW16(surround, ENH_BLOCKL) + 1;
+  max12 = WEBRTC_SPL_MAX(max1, max2);
+  scale = (64 - 31) -
+          WebRtcSpl_CountLeadingZeros64((max12 * max12) * (uint64_t)ENH_BLOCKL);
+  scale=WEBRTC_SPL_MAX(0, scale);
+
+  w00=WebRtcSpl_DotProductWithScale(current,current,ENH_BLOCKL,scale);
+  w11=WebRtcSpl_DotProductWithScale(surround,surround,ENH_BLOCKL,scale);
+  w10=WebRtcSpl_DotProductWithScale(surround,current,ENH_BLOCKL,scale);
+
+  if (w00<0) w00 = WEBRTC_SPL_WORD32_MAX;
+  if (w11<0) w11 = WEBRTC_SPL_WORD32_MAX;
+
+  /* Rescale w00 and w11 to w00prim and w11prim, so that w00prim/w11prim
+     is in Q16 */
+
+  bitsw00 = WebRtcSpl_GetSizeInBits(w00);
+  bitsw11 = WebRtcSpl_GetSizeInBits(w11);
+  bitsw10 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(w10));
+  scale1 = 31 - bitsw00;
+  scale2 = 15 - bitsw11;
+
+  if (scale2>(scale1-16)) {
+    scale2 = scale1 - 16;
+  } else {
+    scale1 = scale2 + 16;
+  }
+
+  w00prim = w00 << scale1;
+  w11prim = (int16_t) WEBRTC_SPL_SHIFT_W32(w11, scale2);
+
+  /* Perform C = sqrt(w11/w00) (C is in Q11 since (16+6)/2=11) */
+  if (w11prim>64) {
+    endiff = WebRtcSpl_DivW32W16(w00prim, w11prim) << 6;
+    C = (int16_t)WebRtcSpl_SqrtFloor(endiff); /* C is in Q11 */
+  } else {
+    C = 1;
+  }
+
+  /* first try enhancement without power-constraint */
+
+  errs = WebRtcIlbcfix_Smooth_odata(odata, current, surround, C);
+
+
+
+  /* if constraint violated by first try, add constraint */
+
+  if ( (6-scale+scale1) > 31) {
+    crit=0;
+  } else {
+    /* crit = 0.05 * w00 (Result in Q-6) */
+    crit = WEBRTC_SPL_SHIFT_W32(
+        WEBRTC_SPL_MUL(ENH_A0, w00prim >> 14),
+        -(6-scale+scale1));
+  }
+
+  if (errs > crit) {
+
+    if( w00 < 1) {
+      w00=1;
+    }
+
+    /* Calculate w11*w00, w10*w10 and w00*w00 in the same Q domain */
+
+    scale1 = bitsw00-15;
+    scale2 = bitsw11-15;
+
+    if (scale2>scale1) {
+      scale = scale2;
+    } else {
+      scale = scale1;
+    }
+
+    w11w00 = (int16_t)WEBRTC_SPL_SHIFT_W32(w11, -scale) *
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale);
+
+    w10w10 = (int16_t)WEBRTC_SPL_SHIFT_W32(w10, -scale) *
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w10, -scale);
+
+    w00w00 = (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale) *
+        (int16_t)WEBRTC_SPL_SHIFT_W32(w00, -scale);
+
+    /* Calculate (w11*w00-w10*w10)/(w00*w00) in Q16 */
+    if (w00w00>65536) {
+      endiff = (w11w00-w10w10);
+      endiff = WEBRTC_SPL_MAX(0, endiff);
+      /* denom is in Q16 */
+      denom = WebRtcSpl_DivW32W16(endiff, (int16_t)(w00w00 >> 16));
+    } else {
+      denom = 65536;
+    }
+
+    if( denom > 7){ /* eliminates numerical problems
+                       for if smooth */
+
+      scale=WebRtcSpl_GetSizeInBits(denom)-15;
+
+      if (scale>0) {
+        /* denomW16 is in Q(16+scale) */
+        denomW16 = (int16_t)(denom >> scale);
+
+        /* num in Q(34-scale) */
+        num = ENH_A0_MINUS_A0A0DIV4 >> scale;
+      } else {
+        /* denomW16 is in Q16 */
+        denomW16=(int16_t)denom;
+
+        /* num in Q34 */
+        num=ENH_A0_MINUS_A0A0DIV4;
+      }
+
+      /* A sqrt( (ENH_A0-(ENH_A0^2)/4)*(w00*w00)/(w11*w00 + w10*w10) ) in Q9 */
+      A = (int16_t)WebRtcSpl_SqrtFloor(WebRtcSpl_DivW32W16(num, denomW16));
+
+      /* B_W32 is in Q30 ( B = 1 - ENH_A0/2 - A * w10/w00 ) */
+      scale1 = 31-bitsw10;
+      scale2 = 21-scale1;
+      w10prim = w10 == 0 ? 0 : w10 * (1 << scale1);
+      w00prim = WEBRTC_SPL_SHIFT_W32(w00, -scale2);
+      scale = bitsw00-scale2-15;
+
+      if (scale>0) {
+        w10prim >>= scale;
+        w00prim >>= scale;
+      }
+
+      if ((w00prim>0)&&(w10prim>0)) {
+        w11_div_w00=WebRtcSpl_DivW32W16(w10prim, (int16_t)w00prim);
+
+        if (WebRtcSpl_GetSizeInBits(w11_div_w00)+WebRtcSpl_GetSizeInBits(A)>31) {
+          B_W32 = 0;
+        } else {
+          B_W32 = (int32_t)1073741824 - (int32_t)ENH_A0DIV2 -
+              WEBRTC_SPL_MUL(A, w11_div_w00);
+        }
+        B = (int16_t)(B_W32 >> 16);  /* B in Q14. */
+      } else {
+        /* No smoothing */
+        A = 0;
+        B = 16384; /* 1 in Q14 */
+      }
+    }
+    else{ /* essentially no difference between cycles;
+             smoothing not needed */
+
+      A = 0;
+      B = 16384; /* 1 in Q14 */
+    }
+
+    /* create smoothed sequence */
+
+    WebRtcSpl_ScaleAndAddVectors(surround, A, 9,
+                                current, B, 14,
+                                odata, ENH_BLOCKL);
+  }
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.h
new file mode 100644
index 0000000000..c8752be64f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * find the smoothed output data
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Smooth(int16_t* odata,   /* (o) smoothed output */
+                          int16_t* current, /* (i) the un enhanced residual for
+                                                     this block */
+                          int16_t* surround /* (i) The approximation from the
+                                                     surrounding sequences */
+                          );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c
new file mode 100644
index 0000000000..9f952bfb93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth_odata.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/smooth_out_data.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "rtc_base/sanitizer.h"
+
+// An s32 + s32 -> s32 addition that's allowed to overflow. (It's still
+// undefined behavior, so not a good idea; this just makes UBSan ignore the
+// violation, so that our old code can continue to do what it's always been
+// doing.)
+static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
+    OverflowingAdd_S32_S32_To_S32(int32_t a, int32_t b) {
+  return a + b;
+}
+
+int32_t WebRtcIlbcfix_Smooth_odata(
+    int16_t *odata,
+    int16_t *psseq,
+    int16_t *surround,
+    int16_t C)
+{
+  int i;
+
+  int16_t err;
+  int32_t errs;
+
+  for(i=0;i<80;i++) {
+    odata[i]= (int16_t)((C * surround[i] + 1024) >> 11);
+  }
+
+  errs=0;
+  for(i=0;i<80;i++) {
+    err = (psseq[i] - odata[i]) >> 3;
+    errs = OverflowingAdd_S32_S32_To_S32(errs, err * err);  // errs in Q-6
+  }
+
+  return errs;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h
new file mode 100644
index 0000000000..318e7b04a2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Smooth_odata.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * help function to WebRtcIlbcfix_Smooth()
+ *---------------------------------------------------------------*/
+
+int32_t WebRtcIlbcfix_Smooth_odata(int16_t* odata,
+                                   int16_t* psseq,
+                                   int16_t* surround,
+                                   int16_t C);
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.c
new file mode 100644
index 0000000000..c3a24750f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.c
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SortSq.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/sort_sq.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  scalar quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SortSq(
+    int16_t *xq,   /* (o) the quantized value */
+    int16_t *index,  /* (o) the quantization index */
+    int16_t x,   /* (i) the value to quantize */
+    const int16_t *cb, /* (i) the quantization codebook */
+    int16_t cb_size  /* (i) the size of the quantization codebook */
+                          ){
+  int i;
+
+  if (x <= cb[0]) {
+    *index = 0;
+    *xq = cb[0];
+  } else {
+    i = 0;
+    while ((x > cb[i]) && (i < (cb_size-1))) {
+      i++;
+    }
+
+    if (x > (((int32_t)cb[i] + cb[i - 1] + 1) >> 1)) {
+      *index = i;
+      *xq = cb[i];
+    } else {
+      *index = i - 1;
+      *xq = cb[i - 1];
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.h
new file mode 100644
index 0000000000..02028dae93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/sort_sq.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SortSq.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  scalar quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SortSq(
+    int16_t* xq,       /* (o) the quantized value */
+    int16_t* index,    /* (o) the quantization index */
+    int16_t x,         /* (i) the value to quantize */
+    const int16_t* cb, /* (i) the quantization codebook */
+    int16_t cb_size    /* (i) the size of the quantization codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.c
new file mode 100644
index 0000000000..c1f04d2287
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.c
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SplitVq.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/split_vq.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/vq3.h"
+#include "modules/audio_coding/codecs/ilbc/vq4.h"
+
+/*----------------------------------------------------------------*
+ *  split vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SplitVq(
+    int16_t *qX,  /* (o) the quantized vector in Q13 */
+    int16_t *index, /* (o) a vector of indexes for all vector
+                                   codebooks in the split */
+    int16_t *X,  /* (i) the vector to quantize */
+    int16_t *CB,  /* (i) the quantizer codebook in Q13 */
+    int16_t *dim, /* (i) the dimension of X and qX */
+    int16_t *cbsize /* (i) the number of vectors in the codebook */
+                           ) {
+
+  int16_t *qXPtr, *indexPtr, *CBPtr, *XPtr;
+
+  /* Quantize X with the 3 vectror quantization tables */
+
+  qXPtr=qX;
+  indexPtr=index;
+  CBPtr=CB;
+  XPtr=X;
+  WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[0]);
+
+  qXPtr+=3;
+  indexPtr+=1;
+  CBPtr+=(dim[0]*cbsize[0]);
+  XPtr+=3;
+  WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[1]);
+
+  qXPtr+=3;
+  indexPtr+=1;
+  CBPtr+=(dim[1]*cbsize[1]);
+  XPtr+=3;
+  WebRtcIlbcfix_Vq4(qXPtr, indexPtr, CBPtr, XPtr, cbsize[2]);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.h
new file mode 100644
index 0000000000..e4b02a2bc2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/split_vq.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SplitVq.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  split vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SplitVq(
+    int16_t* qX,    /* (o) the quantized vector in Q13 */
+    int16_t* index, /* (o) a vector of indexes for all vector
+                                   codebooks in the split */
+    int16_t* X,     /* (i) the vector to quantize */
+    int16_t* CB,    /* (i) the quantizer codebook in Q13 */
+    int16_t* dim,   /* (i) the dimension of X and qX */
+    int16_t* cbsize /* (i) the number of vectors in the codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.c
new file mode 100644
index 0000000000..c58086c03b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.c
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateConstruct.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  decoding of the start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateConstruct(
+    size_t idxForMax,   /* (i) 6-bit index for the quantization of
+                                           max amplitude */
+    int16_t *idxVec,   /* (i) vector of quantization indexes */
+    int16_t *syntDenum,  /* (i) synthesis filter denumerator */
+    int16_t *Out_fix,  /* (o) the decoded state vector */
+    size_t len    /* (i) length of a state vector */
+                                  ) {
+  size_t k;
+  int16_t maxVal;
+  int16_t *tmp1, *tmp2, *tmp3;
+  /* Stack based */
+  int16_t numerator[1+LPC_FILTERORDER];
+  int16_t sampleValVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t sampleMaVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t *sampleVal = &sampleValVec[LPC_FILTERORDER];
+  int16_t *sampleMa = &sampleMaVec[LPC_FILTERORDER];
+  int16_t *sampleAr = &sampleValVec[LPC_FILTERORDER];
+
+  /* initialization of coefficients */
+
+  for (k=0; k<LPC_FILTERORDER+1; k++){
+    numerator[k] = syntDenum[LPC_FILTERORDER-k];
+  }
+
+  /* decoding of the maximum value */
+
+  maxVal = WebRtcIlbcfix_kFrgQuantMod[idxForMax];
+
+  /* decoding of the sample values */
+  tmp1 = sampleVal;
+  tmp2 = &idxVec[len-1];
+
+  if (idxForMax<37) {
+    for(k=0; k<len; k++){
+      /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 2097152 (= 0.5 << 22)
+        maxVal is in Q8 and result is in Q(-1) */
+      *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 2097152) >>
+          22);
+      tmp1++;
+      tmp2--;
+    }
+  } else if (idxForMax<59) {
+    for(k=0; k<len; k++){
+      /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 262144 (= 0.5 << 19)
+        maxVal is in Q5 and result is in Q(-1) */
+      *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 262144) >>
+          19);
+      tmp1++;
+      tmp2--;
+    }
+  } else {
+    for(k=0; k<len; k++){
+      /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 65536 (= 0.5 << 17)
+        maxVal is in Q3 and result is in Q(-1) */
+      *tmp1 = (int16_t)((maxVal * WebRtcIlbcfix_kStateSq3[*tmp2] + 65536) >>
+          17);
+      tmp1++;
+      tmp2--;
+    }
+  }
+
+  /* Set the rest of the data to zero */
+  WebRtcSpl_MemSetW16(&sampleVal[len], 0, len);
+
+  /* circular convolution with all-pass filter */
+
+  /* Set the state to zero */
+  WebRtcSpl_MemSetW16(sampleValVec, 0, (LPC_FILTERORDER));
+
+  /* Run MA filter + AR filter */
+  WebRtcSpl_FilterMAFastQ12(
+      sampleVal, sampleMa,
+      numerator, LPC_FILTERORDER+1, len + LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(&sampleMa[len + LPC_FILTERORDER], 0, (len - LPC_FILTERORDER));
+  WebRtcSpl_FilterARFastQ12(
+      sampleMa, sampleAr,
+      syntDenum, LPC_FILTERORDER+1, 2 * len);
+
+  tmp1 = &sampleAr[len-1];
+  tmp2 = &sampleAr[2*len-1];
+  tmp3 = Out_fix;
+  for(k=0;k<len;k++){
+    (*tmp3) = (*tmp1) + (*tmp2);
+    tmp1--;
+    tmp2--;
+    tmp3++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.h
new file mode 100644
index 0000000000..4c3011937d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_construct.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateConstruct.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Generate the start state from the quantized indexes
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateConstruct(
+    size_t idxForMax,   /* (i) 6-bit index for the quantization of
+                                           max amplitude */
+    int16_t* idxVec,    /* (i) vector of quantization indexes */
+    int16_t* syntDenum, /* (i) synthesis filter denumerator */
+    int16_t* Out_fix,   /* (o) the decoded state vector */
+    size_t len          /* (i) length of a state vector */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.c
new file mode 100644
index 0000000000..7227ac9d45
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.c
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateSearch.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/state_search.h"
+
+#include "modules/audio_coding/codecs/ilbc/abs_quant.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  encoding of start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateSearch(
+    IlbcEncoder *iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
+                               and idxVec, input state_first) */
+    int16_t *residual,   /* (i) target residual vector */
+    int16_t *syntDenum,  /* (i) lpc synthesis filter */
+    int16_t *weightDenum  /* (i) weighting filter denuminator */
+                               ) {
+  size_t k, index;
+  int16_t maxVal;
+  int16_t scale, shift;
+  int32_t maxValsq;
+  int16_t scaleRes;
+  int16_t max;
+  int i;
+  /* Stack based */
+  int16_t numerator[1+LPC_FILTERORDER];
+  int16_t residualLongVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
+  int16_t sampleMa[2*STATE_SHORT_LEN_30MS];
+  int16_t *residualLong = &residualLongVec[LPC_FILTERORDER];
+  int16_t *sampleAr = residualLong;
+
+  /* Scale to maximum 12 bits to avoid saturation in circular convolution filter */
+  max = WebRtcSpl_MaxAbsValueW16(residual, iLBCenc_inst->state_short_len);
+  scaleRes = WebRtcSpl_GetSizeInBits(max)-12;
+  scaleRes = WEBRTC_SPL_MAX(0, scaleRes);
+  /* Set up the filter coefficients for the circular convolution */
+  for (i=0; i<LPC_FILTERORDER+1; i++) {
+    numerator[i] = (syntDenum[LPC_FILTERORDER-i]>>scaleRes);
+  }
+
+  /* Copy the residual to a temporary buffer that we can filter
+   * and set the remaining samples to zero.
+   */
+  WEBRTC_SPL_MEMCPY_W16(residualLong, residual, iLBCenc_inst->state_short_len);
+  WebRtcSpl_MemSetW16(residualLong + iLBCenc_inst->state_short_len, 0, iLBCenc_inst->state_short_len);
+
+  /* Run the Zero-Pole filter (Ciurcular convolution) */
+  WebRtcSpl_MemSetW16(residualLongVec, 0, LPC_FILTERORDER);
+  WebRtcSpl_FilterMAFastQ12(residualLong, sampleMa, numerator,
+                            LPC_FILTERORDER + 1,
+                            iLBCenc_inst->state_short_len + LPC_FILTERORDER);
+  WebRtcSpl_MemSetW16(&sampleMa[iLBCenc_inst->state_short_len + LPC_FILTERORDER], 0, iLBCenc_inst->state_short_len - LPC_FILTERORDER);
+
+  WebRtcSpl_FilterARFastQ12(
+      sampleMa, sampleAr,
+      syntDenum, LPC_FILTERORDER+1, 2 * iLBCenc_inst->state_short_len);
+
+  for(k=0;k<iLBCenc_inst->state_short_len;k++){
+    sampleAr[k] += sampleAr[k+iLBCenc_inst->state_short_len];
+  }
+
+  /* Find maximum absolute value in the vector */
+  maxVal=WebRtcSpl_MaxAbsValueW16(sampleAr, iLBCenc_inst->state_short_len);
+
+  /* Find the best index */
+
+  if ((((int32_t)maxVal)<<scaleRes)<23170) {
+    maxValsq=((int32_t)maxVal*maxVal)<<(2+2*scaleRes);
+  } else {
+    maxValsq=(int32_t)WEBRTC_SPL_WORD32_MAX;
+  }
+
+  index=0;
+  for (i=0;i<63;i++) {
+
+    if (maxValsq>=WebRtcIlbcfix_kChooseFrgQuant[i]) {
+      index=i+1;
+    } else {
+      i=63;
+    }
+  }
+  iLBC_encbits->idxForMax=index;
+
+  /* Rescale the vector before quantization */
+  scale=WebRtcIlbcfix_kScale[index];
+
+  if (index<27) { /* scale table is in Q16, fout[] is in Q(-1) and we want the result to be in Q11 */
+    shift=4;
+  } else { /* scale table is in Q21, fout[] is in Q(-1) and we want the result to be in Q11 */
+    shift=9;
+  }
+
+  /* Set up vectors for AbsQuant and rescale it with the scale factor */
+  WebRtcSpl_ScaleVectorWithSat(sampleAr, sampleAr, scale,
+                              iLBCenc_inst->state_short_len, (int16_t)(shift-scaleRes));
+
+  /* Quantize the values in fout[] */
+  WebRtcIlbcfix_AbsQuant(iLBCenc_inst, iLBC_encbits, sampleAr, weightDenum);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.h
new file mode 100644
index 0000000000..6469138a0e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/state_search.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_StateSearch.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  encoding of start state
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_StateSearch(
+    IlbcEncoder* iLBCenc_inst,
+    /* (i) Encoder instance */
+    iLBC_bits* iLBC_encbits, /* (i/o) Encoded bits (output idxForMax
+                                and idxVec, input state_first) */
+    int16_t* residual,       /* (i) target residual vector */
+    int16_t* syntDenum,      /* (i) lpc synthesis filter */
+    int16_t* weightDenum     /* (i) weighting filter denuminator */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
new file mode 100644
index 0000000000..bbafc1a2ed
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
@@ -0,0 +1,35 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SwapBytes.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
+
+/*----------------------------------------------------------------*
+ * Swap bytes (to simplify operations on Little Endian machines)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SwapBytes(
+    const uint16_t* input,   /* (i) the sequence to swap */
+    size_t wordLength,      /* (i) number or uint16_t to swap */
+    uint16_t* output         /* (o) the swapped sequence */
+                              ) {
+  size_t k;
+  for (k = wordLength; k > 0; k--) {
+    *output++ = (*input >> 8)|(*input << 8);
+    input++;
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
new file mode 100644
index 0000000000..c59bf3068a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
@@ -0,0 +1,35 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_SwapBytes.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * Swap bytes (to simplify operations on Little Endian machines)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_SwapBytes(
+    const uint16_t* input, /* (i) the sequence to swap */
+    size_t wordLength,     /* (i) number or uint16_t to swap */
+    uint16_t* output       /* (o) the swapped sequence */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/empty.cc b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/empty.cc
new file mode 100644
index 0000000000..e69de29bb2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/empty.cc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
new file mode 100644
index 0000000000..e0ca075eda
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
@@ -0,0 +1,238 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+	iLBC Speech Coder ANSI-C Source Code
+
+        iLBC_test.c
+
+******************************************************************/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+/*---------------------------------------------------------------*
+ *  Main program to test iLBC encoding and decoding
+ *
+ *  Usage:
+ *	  exefile_name.exe <infile> <bytefile> <outfile> <channel>
+ *
+ *    <infile>   : Input file, speech for encoder (16-bit pcm file)
+ *    <bytefile> : Bit stream output from the encoder
+ *    <outfile>  : Output file, decoded speech (16-bit pcm file)
+ *    <channel>  : Bit error file, optional (16-bit)
+ *                     1 - Packet received correctly
+ *                     0 - Packet Lost
+ *
+ *--------------------------------------------------------------*/
+
+#define BLOCKL_MAX			240
+#define ILBCNOOFWORDS_MAX	25
+
+
+int main(int argc, char* argv[])
+{
+
+  FILE *ifileid,*efileid,*ofileid, *cfileid;
+  int16_t data[BLOCKL_MAX];
+  uint8_t encoded_data[2 * ILBCNOOFWORDS_MAX];
+  int16_t decoded_data[BLOCKL_MAX];
+  int len_int, mode;
+  short pli;
+  int blockcount = 0;
+  size_t frameLen, len, len_i16s;
+  int16_t speechType;
+  IlbcEncoderInstance *Enc_Inst;
+  IlbcDecoderInstance *Dec_Inst;
+
+#ifdef __ILBC_WITH_40BITACC
+  /* Doublecheck that long long exists */
+  if (sizeof(long)>=sizeof(long long)) {
+    fprintf(stderr, "40-bit simulation is not be supported on this platform\n");
+    exit(0);
+  }
+#endif
+
+  /* get arguments and open files */
+
+  if ((argc!=5) && (argc!=6)) {
+    fprintf(stderr,
+            "\n*-----------------------------------------------*\n");
+    fprintf(stderr,
+            "   %s <20,30> input encoded decoded (channel)\n\n",
+            argv[0]);
+    fprintf(stderr,
+            "   mode    : Frame size for the encoding/decoding\n");
+    fprintf(stderr,
+            "                 20 - 20 ms\n");
+    fprintf(stderr,
+            "                 30 - 30 ms\n");
+    fprintf(stderr,
+            "   input   : Speech for encoder (16-bit pcm file)\n");
+    fprintf(stderr,
+            "   encoded : Encoded bit stream\n");
+    fprintf(stderr,
+            "   decoded : Decoded speech (16-bit pcm file)\n");
+    fprintf(stderr,
+            "   channel : Packet loss pattern, optional (16-bit)\n");
+    fprintf(stderr,
+            "                  1 - Packet received correctly\n");
+    fprintf(stderr,
+            "                  0 - Packet Lost\n");
+    fprintf(stderr,
+            "*-----------------------------------------------*\n\n");
+    exit(1);
+  }
+  mode=atoi(argv[1]);
+  if (mode != 20 && mode != 30) {
+    fprintf(stderr,"Wrong mode %s, must be 20, or 30\n",
+            argv[1]);
+    exit(2);
+  }
+  if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+    exit(2);}
+  if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open encoded file file %s\n",
+            argv[3]); exit(1);}
+  if ( (ofileid=fopen(argv[4],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open decoded file %s\n",
+            argv[4]); exit(1);}
+  if (argc==6) {
+    if( (cfileid=fopen(argv[5],"rb")) == NULL) {
+      fprintf(stderr, "Cannot open channel file %s\n",
+              argv[5]);
+      exit(1);
+    }
+  } else {
+    cfileid=NULL;
+  }
+
+  /* print info */
+
+  fprintf(stderr, "\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*      iLBC test program                            *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,"\nMode           : %2d ms\n", mode);
+  fprintf(stderr,"Input file     : %s\n", argv[2]);
+  fprintf(stderr,"Encoded file   : %s\n", argv[3]);
+  fprintf(stderr,"Output file    : %s\n", argv[4]);
+  if (argc==6) {
+    fprintf(stderr,"Channel file   : %s\n", argv[5]);
+  }
+  fprintf(stderr,"\n");
+
+  /* Create structs */
+  WebRtcIlbcfix_EncoderCreate(&Enc_Inst);
+  WebRtcIlbcfix_DecoderCreate(&Dec_Inst);
+
+
+  /* Initialization */
+
+  WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
+  WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
+  frameLen = (size_t)(mode*8);
+
+  /* loop over input blocks */
+
+  while (fread(data,sizeof(int16_t),frameLen,ifileid) == frameLen) {
+
+    blockcount++;
+
+    /* encoding */
+
+    fprintf(stderr, "--- Encoding block %i --- ",blockcount);
+    len_int = WebRtcIlbcfix_Encode(Enc_Inst, data, frameLen, encoded_data);
+    if (len_int < 0) {
+      fprintf(stderr, "Error encoding\n");
+      exit(0);
+    }
+    len = (size_t)len_int;
+    fprintf(stderr, "\r");
+
+    /* write byte file */
+
+    len_i16s = (len + 1) / sizeof(int16_t);
+    if (fwrite(encoded_data, sizeof(int16_t), len_i16s, efileid) != len_i16s) {
+      return -1;
+    }
+
+    /* get channel data if provided */
+    if (argc==6) {
+      if (fread(&pli, sizeof(int16_t), 1, cfileid)) {
+        if ((pli!=0)&&(pli!=1)) {
+          fprintf(stderr, "Error in channel file\n");
+          exit(0);
+        }
+        if (pli==0) {
+          /* Packet loss -> remove info from frame */
+          memset(encoded_data, 0,
+                 sizeof(int16_t)*ILBCNOOFWORDS_MAX);
+        }
+      } else {
+        fprintf(stderr, "Error. Channel file too short\n");
+        exit(0);
+      }
+    } else {
+      pli=1;
+    }
+
+    /* decoding */
+
+    fprintf(stderr, "--- Decoding block %i --- ",blockcount);
+    if (pli==1) {
+      len_int=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data,
+                                   len, decoded_data,&speechType);
+      if (len_int < 0) {
+        fprintf(stderr, "Error decoding\n");
+        exit(0);
+      }
+      len = (size_t)len_int;
+    } else {
+      len=WebRtcIlbcfix_DecodePlc(Dec_Inst, decoded_data, 1);
+    }
+    fprintf(stderr, "\r");
+
+    /* write output file */
+
+    if (fwrite(decoded_data, sizeof(int16_t), len, ofileid) != len) {
+      return -1;
+    }
+  }
+
+  /* close files */
+
+  fclose(ifileid);  fclose(efileid); fclose(ofileid);
+  if (argc==6) {
+    fclose(cfileid);
+  }
+
+  /* Free structs */
+  WebRtcIlbcfix_EncoderFree(Enc_Inst);
+  WebRtcIlbcfix_DecoderFree(Dec_Inst);
+
+
+  printf("\nDone with simulation\n\n");
+
+  return(0);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
new file mode 100644
index 0000000000..132f3bdb37
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -0,0 +1,215 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+iLBC Speech Coder ANSI-C Source Code
+
+iLBC_test.c
+
+******************************************************************/
+
+#include <math.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <time.h>
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+//#define JUNK_DATA
+#ifdef JUNK_DATA
+#define SEED_FILE "randseed.txt"
+#endif
+
+
+/*----------------------------------------------------------------*
+*  Main program to test iLBC encoding and decoding
+*
+*  Usage:
+*		exefile_name.exe <infile> <bytefile> <outfile>
+*
+*---------------------------------------------------------------*/
+
+int main(int argc, char* argv[])
+{
+  FILE *ifileid,*efileid,*ofileid, *chfileid;
+  short encoded_data[55], data[240], speechType;
+  int len_int, mode;
+  short pli;
+  size_t len, readlen;
+  int blockcount = 0;
+
+  IlbcEncoderInstance *Enc_Inst;
+  IlbcDecoderInstance *Dec_Inst;
+#ifdef JUNK_DATA
+  size_t i;
+  FILE *seedfile;
+  unsigned int random_seed = (unsigned int) time(NULL);//1196764538
+#endif
+
+  /* Create structs */
+  WebRtcIlbcfix_EncoderCreate(&Enc_Inst);
+  WebRtcIlbcfix_DecoderCreate(&Dec_Inst);
+
+  /* get arguments and open files */
+
+  if (argc != 6 ) {
+    fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n",
+            argv[0]);
+    fprintf(stderr, "Example:\n");
+    fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]);
+    exit(1);
+  }
+  mode=atoi(argv[1]);
+  if (mode != 20 && mode != 30) {
+    fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]);
+    exit(2);
+  }
+  if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+    exit(2);}
+  if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open channelfile file %s\n",
+            argv[3]); exit(3);}
+  if( (ofileid=fopen(argv[4],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open output file %s\n",
+            argv[4]); exit(3);}
+  if ( (chfileid=fopen(argv[5],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open channel file file %s\n", argv[5]);
+    exit(2);
+  }
+  /* print info */
+  fprintf(stderr, "\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*      iLBCtest                                     *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+		"*---------------------------------------------------*\n");
+#ifdef SPLIT_10MS
+  fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode);
+#else
+  fprintf(stderr,"\nMode          : %2d ms\n", mode);
+#endif
+  fprintf(stderr,"\nInput file    : %s\n", argv[2]);
+  fprintf(stderr,"Coded file    : %s\n", argv[3]);
+  fprintf(stderr,"Output file   : %s\n\n", argv[4]);
+  fprintf(stderr,"Channel file  : %s\n\n", argv[5]);
+
+#ifdef JUNK_DATA
+  srand(random_seed);
+
+  if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+    fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+  }
+  else {
+    fprintf(seedfile, "%u\n", random_seed);
+    fclose(seedfile);
+  }
+#endif
+
+  /* Initialization */
+  WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
+  WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
+
+  /* loop over input blocks */
+#ifdef SPLIT_10MS
+  readlen = 80;
+#else
+  readlen = (size_t)(mode << 3);
+#endif
+  while(fread(data, sizeof(short), readlen, ifileid) == readlen) {
+    blockcount++;
+
+    /* encoding */
+    fprintf(stderr, "--- Encoding block %i --- ",blockcount);
+    len_int=WebRtcIlbcfix_Encode(Enc_Inst, data, readlen, encoded_data);
+    if (len_int < 0) {
+      fprintf(stderr, "Error encoding\n");
+      exit(0);
+    }
+    len = (size_t)len_int;
+    fprintf(stderr, "\r");
+
+#ifdef JUNK_DATA
+    for ( i = 0; i < len; i++) {
+      encoded_data[i] = (short) (encoded_data[i] + (short) rand());
+    }
+#endif
+    /* write byte file */
+    if(len != 0){ //len may be 0 in 10ms split case
+      fwrite(encoded_data,1,len,efileid);
+
+      /* get channel data if provided */
+      if (argc==6) {
+        if (fread(&pli, sizeof(int16_t), 1, chfileid)) {
+          if ((pli!=0)&&(pli!=1)) {
+            fprintf(stderr, "Error in channel file\n");
+            exit(0);
+          }
+          if (pli==0) {
+            /* Packet loss -> remove info from frame */
+            memset(encoded_data, 0, sizeof(int16_t)*25);
+          }
+        } else {
+          fprintf(stderr, "Error. Channel file too short\n");
+          exit(0);
+        }
+      } else {
+        pli=1;
+      }
+
+      /* decoding */
+      fprintf(stderr, "--- Decoding block %i --- ",blockcount);
+      if (pli==1) {
+        len_int = WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data,
+                                       &speechType);
+        if (len_int < 0) {
+          fprintf(stderr, "Error decoding\n");
+          exit(0);
+        }
+        len = (size_t)len_int;
+      } else {
+        len=WebRtcIlbcfix_DecodePlc(Dec_Inst, data, 1);
+      }
+      fprintf(stderr, "\r");
+
+      /* write output file */
+      fwrite(data,sizeof(short),len,ofileid);
+    }
+  }
+
+#ifdef JUNK_DATA
+  if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
+    fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
+  }
+  else {
+    fprintf(seedfile, "ok\n\n");
+    fclose(seedfile);
+  }
+#endif
+
+  /* free structs */
+  WebRtcIlbcfix_EncoderFree(Enc_Inst);
+  WebRtcIlbcfix_DecoderFree(Dec_Inst);
+
+  /* close files */
+  fclose(ifileid);
+  fclose(efileid);
+  fclose(ofileid);
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
new file mode 100644
index 0000000000..a62a42edf6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
@@ -0,0 +1,343 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+	iLBC Speech Coder ANSI-C Source Code
+
+        iLBC_test.c
+
+******************************************************************/
+
+#include <math.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/nit_encode.h"
+#include "modules/audio_coding/codecs/ilbc/encode.h"
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "modules/audio_coding/codecs/ilbc/decode.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+
+#define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS)/2
+
+/* Runtime statistics */
+#include <time.h>
+/* #define CLOCKS_PER_SEC  1000 */
+
+/*----------------------------------------------------------------*
+ *  Encoder interface function
+ *---------------------------------------------------------------*/
+
+short encode(                         /* (o) Number of bytes encoded */
+    IlbcEncoder *iLBCenc_inst,    /* (i/o) Encoder instance */
+    int16_t *encoded_data,      /* (o) The encoded bytes */
+    int16_t *data               /* (i) The signal block to encode */
+                                                        ){
+
+  /* do the actual encoding */
+  WebRtcIlbcfix_Encode((uint16_t *)encoded_data, data, iLBCenc_inst);
+
+  return (iLBCenc_inst->no_of_bytes);
+}
+
+/*----------------------------------------------------------------*
+ *  Decoder interface function
+ *---------------------------------------------------------------*/
+
+short decode( /* (o) Number of decoded samples */
+    IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */
+    short *decoded_data, /* (o) Decoded signal block */
+    short *encoded_data, /* (i) Encoded bytes */
+    short mode           /* (i) 0=PL, 1=Normal */
+              ){
+
+  /* check if mode is valid */
+
+  if (mode<0 || mode>1) {
+    printf("\nERROR - Wrong mode - 0, 1 allowed\n"); exit(3);}
+
+  /* do actual decoding of block */
+
+  WebRtcIlbcfix_Decode(decoded_data, (uint16_t *)encoded_data,
+                       iLBCdec_inst, mode);
+
+  return (iLBCdec_inst->blockl);
+}
+
+/*----------------------------------------------------------------*
+ *  Main program to test iLBC encoding and decoding
+ *
+ *  Usage:
+ *		exefile_name.exe <infile> <bytefile> <outfile> <channelfile>
+ *
+ *---------------------------------------------------------------*/
+
+#define MAXFRAMES   10000
+#define MAXFILELEN (BLOCKL_MAX*MAXFRAMES)
+
+int main(int argc, char* argv[])
+{
+
+  /* Runtime statistics */
+
+  float starttime1, starttime2;
+  float runtime1, runtime2;
+  float outtime;
+
+  FILE *ifileid,*efileid,*ofileid, *chfileid;
+  short *inputdata, *encodeddata, *decodeddata;
+  short *channeldata;
+  int blockcount = 0, noOfBlocks=0, i, noOfLostBlocks=0;
+  short mode;
+  IlbcEncoder Enc_Inst;
+  IlbcDecoder Dec_Inst;
+
+  short frameLen;
+  short count;
+#ifdef SPLIT_10MS
+  short size;
+#endif
+
+  inputdata=(short*) malloc(MAXFILELEN*sizeof(short));
+  if (inputdata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    exit(0);
+  }
+  encodeddata=(short*) malloc(ILBCNOOFWORDS_MAX*MAXFRAMES*sizeof(short));
+  if (encodeddata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    free(inputdata);
+    exit(0);
+  }
+  decodeddata=(short*) malloc(MAXFILELEN*sizeof(short));
+  if (decodeddata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    free(inputdata);
+    free(encodeddata);
+    exit(0);
+  }
+  channeldata=(short*) malloc(MAXFRAMES*sizeof(short));
+  if (channeldata==NULL) {
+    fprintf(stderr,"Could not allocate memory for vector\n");
+    free(inputdata);
+    free(encodeddata);
+    free(decodeddata);
+    exit(0);
+  }
+
+  /* get arguments and open files */
+
+  if (argc != 6 ) {
+    fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n",
+            argv[0]);
+    fprintf(stderr, "Example:\n");
+    fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]);
+    exit(1);
+  }
+  mode=atoi(argv[1]);
+  if (mode != 20 && mode != 30) {
+    fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]);
+    exit(2);
+  }
+  if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open input file %s\n", argv[2]);
+    exit(2);}
+  if ( (efileid=fopen(argv[3],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open channelfile file %s\n",
+            argv[3]); exit(3);}
+  if( (ofileid=fopen(argv[4],"wb")) == NULL) {
+    fprintf(stderr, "Cannot open output file %s\n",
+            argv[4]); exit(3);}
+  if ( (chfileid=fopen(argv[5],"rb")) == NULL) {
+    fprintf(stderr,"Cannot open channel file file %s\n", argv[5]);
+    exit(2);}
+
+
+  /* print info */
+#ifndef PRINT_MIPS
+  fprintf(stderr, "\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*      iLBCtest                                     *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*                                                   *\n");
+  fprintf(stderr,
+          "*---------------------------------------------------*\n");
+#ifdef SPLIT_10MS
+  fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode);
+#else
+  fprintf(stderr,"\nMode          : %2d ms\n", mode);
+#endif
+  fprintf(stderr,"\nInput file    : %s\n", argv[2]);
+  fprintf(stderr,"Coded file    : %s\n", argv[3]);
+  fprintf(stderr,"Output file   : %s\n\n", argv[4]);
+  fprintf(stderr,"Channel file  : %s\n\n", argv[5]);
+#endif
+
+  /* Initialization */
+
+  WebRtcIlbcfix_EncoderInit(&Enc_Inst, mode);
+  WebRtcIlbcfix_DecoderInit(&Dec_Inst, mode, 1);
+
+  /* extract the input file and channel file */
+
+#ifdef SPLIT_10MS
+  frameLen = (mode==20)? 80:160;
+  fread(Enc_Inst.past_samples, sizeof(short), frameLen, ifileid);
+  Enc_Inst.section = 0;
+
+  while( fread(&inputdata[noOfBlocks*80], sizeof(short),
+               80, ifileid) == 80 ) {
+    noOfBlocks++;
+  }
+
+  noOfBlocks += frameLen/80;
+  frameLen = 80;
+#else
+  frameLen = Enc_Inst.blockl;
+
+  while( fread(&inputdata[noOfBlocks*Enc_Inst.blockl],sizeof(short),
+               Enc_Inst.blockl,ifileid)==(uint16_t)Enc_Inst.blockl){
+    noOfBlocks++;
+  }
+#endif
+
+
+  while ((fread(&channeldata[blockcount],sizeof(short), 1,chfileid)==1)
+            && ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) )) {
+    blockcount++;
+  }
+
+  if ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) ) {
+    fprintf(stderr,"Channel file %s is too short\n", argv[4]);
+    free(inputdata);
+    free(encodeddata);
+    free(decodeddata);
+    free(channeldata);
+    exit(0);
+  }
+
+  count=0;
+
+  /* Runtime statistics */
+
+  starttime1 = clock()/(float)CLOCKS_PER_SEC;
+
+  /* Encoding loop */
+#ifdef PRINT_MIPS
+  printf("-1 -1\n");
+#endif
+
+#ifdef SPLIT_10MS
+  /* "Enc_Inst.section != 0" is to make sure we run through full
+     lengths of all vectors for 10ms split mode.
+  */
+  //   while( (count < noOfBlocks) || (Enc_Inst.section != 0) )    {
+  while( count < blockcount * (Enc_Inst.blockl/frameLen) )    {
+
+    encode(&Enc_Inst, &encodeddata[Enc_Inst.no_of_words *
+                                   (count/(Enc_Inst.nsub/2))],
+           &inputdata[frameLen * count] );
+#else
+    while (count < noOfBlocks) {
+      encode( &Enc_Inst, &encodeddata[Enc_Inst.no_of_words * count],
+              &inputdata[frameLen * count] );
+#endif
+
+#ifdef PRINT_MIPS
+      printf("-1 -1\n");
+#endif
+
+      count++;
+    }
+
+    count=0;
+
+    /* Runtime statistics */
+
+    starttime2=clock()/(float)CLOCKS_PER_SEC;
+    runtime1 = (float)(starttime2-starttime1);
+
+    /* Decoding loop */
+
+    while (count < blockcount) {
+      if (channeldata[count]==1) {
+        /* Normal decoding */
+        decode(&Dec_Inst, &decodeddata[count * Dec_Inst.blockl],
+               &encodeddata[Dec_Inst.no_of_words * count], 1);
+      } else if (channeldata[count]==0) {
+        /* PLC */
+        short emptydata[ILBCNOOFWORDS_MAX];
+        memset(emptydata, 0, Dec_Inst.no_of_words*sizeof(short));
+        decode(&Dec_Inst, &decodeddata[count*Dec_Inst.blockl],
+               emptydata, 0);
+        noOfLostBlocks++;
+      } else {
+        printf("Error in channel file (values have to be either 1 or 0)\n");
+        exit(0);
+      }
+#ifdef PRINT_MIPS
+      printf("-1 -1\n");
+#endif
+
+      count++;
+    }
+
+    /* Runtime statistics */
+
+    runtime2 = (float)(clock()/(float)CLOCKS_PER_SEC-starttime2);
+
+    outtime = (float)((float)blockcount*
+                      (float)mode/1000.0);
+
+#ifndef PRINT_MIPS
+    printf("\nLength of speech file: %.1f s\n", outtime);
+    printf("Lost frames          : %.1f%%\n\n", 100*(float)noOfLostBlocks/(float)blockcount);
+
+    printf("Time to run iLBC_encode+iLBC_decode:");
+    printf(" %.1f s (%.1f%% of realtime)\n", runtime1+runtime2,
+           (100*(runtime1+runtime2)/outtime));
+
+    printf("Time in iLBC_encode                :");
+    printf(" %.1f s (%.1f%% of total runtime)\n",
+           runtime1, 100.0*runtime1/(runtime1+runtime2));
+
+    printf("Time in iLBC_decode                :");
+    printf(" %.1f s (%.1f%% of total runtime)\n\n",
+           runtime2, 100.0*runtime2/(runtime1+runtime2));
+#endif
+
+    /* Write data to files */
+    for (i=0; i<blockcount; i++) {
+      fwrite(&encodeddata[i*Enc_Inst.no_of_words], sizeof(short),
+             Enc_Inst.no_of_words, efileid);
+    }
+    for (i=0;i<blockcount;i++) {
+      fwrite(&decodeddata[i*Enc_Inst.blockl],sizeof(short),Enc_Inst.blockl,ofileid);
+    }
+
+    /* return memory and close files */
+
+    free(inputdata);
+    free(encodeddata);
+    free(decodeddata);
+    free(channeldata);
+    fclose(ifileid);  fclose(efileid); fclose(ofileid);
+    return(0);
+  }
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c
new file mode 100644
index 0000000000..a9a0147b9d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c
@@ -0,0 +1,241 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_UnpackBits.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+int16_t WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
+    const uint16_t *bitstream,    /* (i) The packatized bitstream */
+    iLBC_bits *enc_bits,  /* (o) Paramerers from bitstream */
+    int16_t mode     /* (i) Codec mode (20 or 30) */
+                                        ) {
+  const uint16_t *bitstreamPtr;
+  int i, k;
+  int16_t *tmpPtr;
+
+  bitstreamPtr=bitstream;
+
+  /* First int16_t */
+  enc_bits->lsf[0]  =  (*bitstreamPtr)>>10;       /* Bit 0..5  */
+  enc_bits->lsf[1]  = ((*bitstreamPtr)>>3)&0x7F;      /* Bit 6..12 */
+  enc_bits->lsf[2]  = ((*bitstreamPtr)&0x7)<<4;      /* Bit 13..15 */
+  bitstreamPtr++;
+  /* Second int16_t */
+  enc_bits->lsf[2] |= ((*bitstreamPtr)>>12)&0xF;      /* Bit 0..3  */
+
+  if (mode==20) {
+    enc_bits->startIdx             = ((*bitstreamPtr)>>10)&0x3;  /* Bit 4..5  */
+    enc_bits->state_first          = ((*bitstreamPtr)>>9)&0x1;  /* Bit 6  */
+    enc_bits->idxForMax            = ((*bitstreamPtr)>>3)&0x3F;  /* Bit 7..12 */
+    enc_bits->cb_index[0]          = ((*bitstreamPtr)&0x7)<<4;  /* Bit 13..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>12)&0xE;  /* Bit 0..2  */
+    enc_bits->gain_index[0]        = ((*bitstreamPtr)>>8)&0x18;  /* Bit 3..4  */
+    enc_bits->gain_index[1]        = ((*bitstreamPtr)>>7)&0x8;  /* Bit 5  */
+    enc_bits->cb_index[3]          = ((*bitstreamPtr)>>2)&0xFE;  /* Bit 6..12 */
+    enc_bits->gain_index[3]        = ((*bitstreamPtr)<<2)&0x10;  /* Bit 13  */
+    enc_bits->gain_index[4]        = ((*bitstreamPtr)<<2)&0x8;  /* Bit 14  */
+    enc_bits->gain_index[6]        = ((*bitstreamPtr)<<4)&0x10;  /* Bit 15  */
+  } else { /* mode==30 */
+    enc_bits->lsf[3]               = ((*bitstreamPtr)>>6)&0x3F;  /* Bit 4..9  */
+    enc_bits->lsf[4]               = ((*bitstreamPtr)<<1)&0x7E;  /* Bit 10..15 */
+    bitstreamPtr++;
+    /* Third int16_t */
+    enc_bits->lsf[4]              |= ((*bitstreamPtr)>>15)&0x1;  /* Bit 0  */
+    enc_bits->lsf[5]               = ((*bitstreamPtr)>>8)&0x7F;  /* Bit 1..7  */
+    enc_bits->startIdx             = ((*bitstreamPtr)>>5)&0x7;  /* Bit 8..10 */
+    enc_bits->state_first          = ((*bitstreamPtr)>>4)&0x1;  /* Bit 11  */
+    enc_bits->idxForMax            = ((*bitstreamPtr)<<2)&0x3C;  /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 4:th int16_t */
+    enc_bits->idxForMax           |= ((*bitstreamPtr)>>14)&0x3;  /* Bit 0..1  */
+    enc_bits->cb_index[0]        = ((*bitstreamPtr)>>7)&0x78;  /* Bit 2..5  */
+    enc_bits->gain_index[0]        = ((*bitstreamPtr)>>5)&0x10;  /* Bit 6  */
+    enc_bits->gain_index[1]        = ((*bitstreamPtr)>>5)&0x8;  /* Bit 7  */
+    enc_bits->cb_index[3]          = ((*bitstreamPtr))&0xFC;  /* Bit 8..13 */
+    enc_bits->gain_index[3]        = ((*bitstreamPtr)<<3)&0x10;  /* Bit 14  */
+    enc_bits->gain_index[4]        = ((*bitstreamPtr)<<3)&0x8;  /* Bit 15  */
+  }
+  /* Class 2 bits of ULP */
+  /* 4:th to 6:th int16_t for 20 ms case
+     5:th to 7:th int16_t for 30 ms case */
+  bitstreamPtr++;
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<3; k++) {
+    for (i=15; i>=0; i--) {
+      (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 7:th int16_t */
+    for (i=15; i>6; i--) {
+      (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>4)&0x4; /* Bit 9  */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>1)&0x4; /* Bit 12  */
+    enc_bits->gain_index[6]       |= ((*bitstreamPtr)<<1)&0x8; /* Bit 13  */
+    enc_bits->gain_index[7]        = ((*bitstreamPtr)<<2)&0xC; /* Bit 14..15 */
+
+  } else { /* mode==30 */
+    /* 8:th int16_t */
+    for (i=15; i>5; i--) {
+      (*tmpPtr)                  = (((*bitstreamPtr)>>i)<<2)&0x4;
+      /* Bit 15-i  */
+      tmpPtr++;
+    }
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>3)&0x6; /* Bit 10..11 */
+    enc_bits->gain_index[0]       |= ((*bitstreamPtr))&0x8;  /* Bit 12  */
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr))&0x4;  /* Bit 13  */
+    enc_bits->cb_index[3]         |= ((*bitstreamPtr))&0x2;  /* Bit 14  */
+    enc_bits->cb_index[6]          = ((*bitstreamPtr)<<7)&0x80; /* Bit 15  */
+    bitstreamPtr++;
+    /* 9:th int16_t */
+    enc_bits->cb_index[6]         |= ((*bitstreamPtr)>>9)&0x7E; /* Bit 0..5  */
+    enc_bits->cb_index[9]          = ((*bitstreamPtr)>>2)&0xFE; /* Bit 6..12 */
+    enc_bits->cb_index[12]         = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 10:th int16_t */
+    enc_bits->cb_index[12]         |= ((*bitstreamPtr)>>11)&0x1E;/* Bit 0..3 */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>8)&0xC; /* Bit 4..5  */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>7)&0x6; /* Bit 6..7  */
+    enc_bits->gain_index[6]        = ((*bitstreamPtr)>>3)&0x18; /* Bit 8..9  */
+    enc_bits->gain_index[7]        = ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */
+    enc_bits->gain_index[9]        = ((*bitstreamPtr)<<1)&0x10; /* Bit 12  */
+    enc_bits->gain_index[10]       = ((*bitstreamPtr)<<1)&0x8; /* Bit 13  */
+    enc_bits->gain_index[12]       = ((*bitstreamPtr)<<3)&0x10; /* Bit 14  */
+    enc_bits->gain_index[13]       = ((*bitstreamPtr)<<3)&0x8; /* Bit 15  */
+  }
+  bitstreamPtr++;
+  /* Class 3 bits of ULP */
+  /*  8:th to 14:th int16_t for 20 ms case
+      11:th to 17:th int16_t for 30 ms case */
+  tmpPtr=enc_bits->idxVec;
+  for (k=0; k<7; k++) {
+    for (i=14; i>=0; i-=2) {
+      (*tmpPtr)                 |= ((*bitstreamPtr)>>i)&0x3; /* Bit 15-i..14-i*/
+      tmpPtr++;
+    }
+    bitstreamPtr++;
+  }
+
+  if (mode==20) {
+    /* 15:th int16_t */
+    enc_bits->idxVec[56]          |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>13)&0x1; /* Bit 2  */
+    enc_bits->cb_index[1]          = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9  */
+    enc_bits->cb_index[2]          = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 16:th int16_t */
+    enc_bits->cb_index[2]         |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0  */
+    enc_bits->gain_index[0]       |= ((*bitstreamPtr)>>12)&0x7; /* Bit 1..3  */
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>10)&0x3; /* Bit 4..5  */
+    enc_bits->gain_index[2]        = ((*bitstreamPtr)>>7)&0x7; /* Bit 6..8  */
+    enc_bits->cb_index[3]         |= ((*bitstreamPtr)>>6)&0x1; /* Bit 9  */
+    enc_bits->cb_index[4]          = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
+    bitstreamPtr++;
+    /* 17:th int16_t */
+    enc_bits->cb_index[4]         |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0  */
+    enc_bits->cb_index[5]          = ((*bitstreamPtr)>>8)&0x7F; /* Bit 1..7  */
+    enc_bits->cb_index[6]          = ((*bitstreamPtr))&0xFF; /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 18:th int16_t */
+    enc_bits->cb_index[7]          = (*bitstreamPtr)>>8;  /* Bit 0..7  */
+    enc_bits->cb_index[8]          = (*bitstreamPtr)&0xFF;  /* Bit 8..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3  */
+    enc_bits->gain_index[5]        = ((*bitstreamPtr)>>9)&0x7; /* Bit 4..6  */
+    enc_bits->gain_index[6]       |= ((*bitstreamPtr)>>6)&0x7; /* Bit 7..9  */
+    enc_bits->gain_index[7]       |= ((*bitstreamPtr)>>4)&0x3; /* Bit 10..11 */
+    enc_bits->gain_index[8]        = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
+  } else { /* mode==30 */
+    /* 18:th int16_t */
+    enc_bits->idxVec[56]          |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->idxVec[57]          |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3  */
+    enc_bits->cb_index[0]         |= ((*bitstreamPtr)>>11)&1; /* Bit 4  */
+    enc_bits->cb_index[1]          = ((*bitstreamPtr)>>4)&0x7F; /* Bit 5..11 */
+    enc_bits->cb_index[2]          = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 19:th int16_t */
+    enc_bits->cb_index[2]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->gain_index[0]       |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5  */
+    enc_bits->gain_index[1]       |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7  */
+    enc_bits->gain_index[2]        = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
+    enc_bits->cb_index[3]         |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11  */
+    enc_bits->cb_index[4]          = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 20:th int16_t */
+    enc_bits->cb_index[4]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->cb_index[5]          = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9  */
+    enc_bits->cb_index[6]         |= ((*bitstreamPtr)>>5)&0x1; /* Bit 10  */
+    enc_bits->cb_index[7]          = ((*bitstreamPtr)<<3)&0xF8; /* Bit 11..15 */
+    bitstreamPtr++;
+    /* 21:st int16_t */
+    enc_bits->cb_index[7]         |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->cb_index[8]          = ((*bitstreamPtr)>>5)&0xFF; /* Bit 3..10 */
+    enc_bits->cb_index[9]         |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11  */
+    enc_bits->cb_index[10]         = ((*bitstreamPtr)<<4)&0xF0; /* Bit 12..15 */
+    bitstreamPtr++;
+    /* 22:nd int16_t */
+    enc_bits->cb_index[10]        |= ((*bitstreamPtr)>>12)&0xF; /* Bit 0..3  */
+    enc_bits->cb_index[11]         = ((*bitstreamPtr)>>4)&0xFF; /* Bit 4..11 */
+    enc_bits->cb_index[12]        |= ((*bitstreamPtr)>>3)&0x1; /* Bit 12  */
+    enc_bits->cb_index[13]         = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
+    bitstreamPtr++;
+    /* 23:rd int16_t */
+    enc_bits->cb_index[13]        |= ((*bitstreamPtr)>>11)&0x1F;/* Bit 0..4  */
+    enc_bits->cb_index[14]         = ((*bitstreamPtr)>>3)&0xFF; /* Bit 5..12 */
+    enc_bits->gain_index[3]       |= ((*bitstreamPtr)>>1)&0x3; /* Bit 13..14 */
+    enc_bits->gain_index[4]       |= ((*bitstreamPtr)&0x1);  /* Bit 15  */
+    bitstreamPtr++;
+    /* 24:rd int16_t */
+    enc_bits->gain_index[5]        = ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2  */
+    enc_bits->gain_index[6]       |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5  */
+    enc_bits->gain_index[7]       |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7  */
+    enc_bits->gain_index[8]        = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
+    enc_bits->gain_index[9]       |= ((*bitstreamPtr)>>1)&0xF; /* Bit 11..14 */
+    enc_bits->gain_index[10]      |= ((*bitstreamPtr)<<2)&0x4; /* Bit 15  */
+    bitstreamPtr++;
+    /* 25:rd int16_t */
+    enc_bits->gain_index[10]      |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1  */
+    enc_bits->gain_index[11]       = ((*bitstreamPtr)>>11)&0x7; /* Bit 2..4  */
+    enc_bits->gain_index[12]      |= ((*bitstreamPtr)>>7)&0xF; /* Bit 5..8  */
+    enc_bits->gain_index[13]      |= ((*bitstreamPtr)>>4)&0x7; /* Bit 9..11 */
+    enc_bits->gain_index[14]       = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
+  }
+  /* Last bit should be zero, otherwise it's an "empty" frame */
+  if (((*bitstreamPtr)&0x1) == 1) {
+    return(1);
+  } else {
+    return(0);
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h
new file mode 100644
index 0000000000..1a63280e6b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h
@@ -0,0 +1,39 @@
+/*
+ *  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.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_UnpackBits.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
+
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  unpacking of bits from bitstream, i.e., vector of bytes
+ *---------------------------------------------------------------*/
+
+int16_t
+WebRtcIlbcfix_UnpackBits(/* (o) "Empty" frame indicator */
+                         const uint16_t*
+                             bitstream, /* (i) The packatized bitstream */
+                         iLBC_bits*
+                             enc_bits, /* (o) Paramerers from bitstream */
+                         int16_t mode  /* (i) Codec mode (20 or 30) */
+                         );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.c
new file mode 100644
index 0000000000..d9375fb995
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.c
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq3.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/vq3.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+/*----------------------------------------------------------------*
+ *  vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq3(
+    int16_t *Xq, /* quantized vector (Q13) */
+    int16_t *index,
+    int16_t *CB, /* codebook in Q13 */
+    int16_t *X,  /* vector to quantize (Q13) */
+    int16_t n_cb
+                       ){
+  int16_t i, j;
+  int16_t pos, minindex=0;
+  int16_t tmp;
+  int32_t dist, mindist;
+
+  pos = 0;
+  mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
+
+  /* Find the codebook with the lowest square distance */
+  for (j = 0; j < n_cb; j++) {
+    tmp = X[0] - CB[pos];
+    dist = tmp * tmp;
+    for (i = 1; i < 3; i++) {
+      tmp = X[i] - CB[pos + i];
+      dist += tmp * tmp;
+    }
+
+    if (dist < mindist) {
+      mindist = dist;
+      minindex = j;
+    }
+    pos += 3;
+  }
+
+  /* Store the quantized codebook and the index */
+  for (i = 0; i < 3; i++) {
+    Xq[i] = CB[minindex*3 + i];
+  }
+  *index = minindex;
+
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.h
new file mode 100644
index 0000000000..c946478a1a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq3.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq3.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Vector quantization of order 3 (based on MSE)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq3(
+    int16_t* Xq,    /* (o) the quantized vector (Q13) */
+    int16_t* index, /* (o) the quantization index */
+    int16_t* CB,    /* (i) the vector quantization codebook (Q13) */
+    int16_t* X,     /* (i) the vector to quantize (Q13) */
+    int16_t n_cb    /* (i) the number of vectors in the codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.c
new file mode 100644
index 0000000000..c9a65aec2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.c
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq4.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/vq4.h"
+
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+
+/*----------------------------------------------------------------*
+ *  vector quantization
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq4(
+    int16_t *Xq, /* quantized vector (Q13) */
+    int16_t *index,
+    int16_t *CB, /* codebook in Q13 */
+    int16_t *X,  /* vector to quantize (Q13) */
+    int16_t n_cb
+                       ){
+  int16_t i, j;
+  int16_t pos, minindex=0;
+  int16_t tmp;
+  int32_t dist, mindist;
+
+  pos = 0;
+  mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
+
+  /* Find the codebook with the lowest square distance */
+  for (j = 0; j < n_cb; j++) {
+    tmp = X[0] - CB[pos];
+    dist = tmp * tmp;
+    for (i = 1; i < 4; i++) {
+      tmp = X[i] - CB[pos + i];
+      dist += tmp * tmp;
+    }
+
+    if (dist < mindist) {
+      mindist = dist;
+      minindex = j;
+    }
+    pos += 4;
+  }
+
+  /* Store the quantized codebook and the index */
+  for (i = 0; i < 4; i++) {
+    Xq[i] = CB[minindex*4 + i];
+  }
+  *index = minindex;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.h
new file mode 100644
index 0000000000..6d14830c03
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/vq4.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Vq4.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
+
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  Vector quantization of order 4 (based on MSE)
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Vq4(
+    int16_t* Xq,    /* (o) the quantized vector (Q13) */
+    int16_t* index, /* (o) the quantization index */
+    int16_t* CB,    /* (i) the vector quantization codebook (Q13) */
+    int16_t* X,     /* (i) the vector to quantize (Q13) */
+    int16_t n_cb    /* (i) the number of vectors in the codebook */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
new file mode 100644
index 0000000000..e82d167220
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Window32W32.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/window32_w32.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ *  window multiplication
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Window32W32(
+    int32_t *z,    /* Output */
+    int32_t *x,    /* Input (same domain as Output)*/
+    const int32_t  *y,  /* Q31 Window */
+    size_t N     /* length to process */
+                               ) {
+  size_t i;
+  int16_t x_low, x_hi, y_low, y_hi;
+  int16_t left_shifts;
+  int32_t temp;
+
+  left_shifts = (int16_t)WebRtcSpl_NormW32(x[0]);
+  WebRtcSpl_VectorBitShiftW32(x, N, x, (int16_t)(-left_shifts));
+
+
+  /* The double precision numbers use a special representation:
+   * w32 = hi<<16 + lo<<1
+   */
+  for (i = 0; i < N; i++) {
+    /* Extract higher bytes */
+    x_hi = (int16_t)(x[i] >> 16);
+    y_hi = (int16_t)(y[i] >> 16);
+
+    /* Extract lower bytes, defined as (w32 - hi<<16)>>1 */
+    x_low = (int16_t)((x[i] - (x_hi << 16)) >> 1);
+
+    y_low = (int16_t)((y[i] - (y_hi << 16)) >> 1);
+
+    /* Calculate z by a 32 bit multiplication using both low and high from x and y */
+    temp = ((x_hi * y_hi) << 1) + ((x_hi * y_low) >> 14);
+
+    z[i] = temp + ((x_low * y_hi) >> 14);
+  }
+
+  WebRtcSpl_VectorBitShiftW32(z, N, z, left_shifts);
+
+  return;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
new file mode 100644
index 0000000000..15d72c5ba2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_Window32W32.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ *  window multiplication
+ *---------------------------------------------------------------*/
+
+void WebRtcIlbcfix_Window32W32(int32_t* z, /* Output */
+                               int32_t* x, /* Input (same domain as Output)*/
+                               const int32_t* y, /* Q31 Window */
+                               size_t N          /* length to process */
+                               );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
new file mode 100644
index 0000000000..9dc880b37e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
@@ -0,0 +1,142 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_XcorrCoef.c
+
+******************************************************************/
+
+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
+
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+
+/*----------------------------------------------------------------*
+ * cross correlation which finds the optimal lag for the
+ * crossCorr*crossCorr/(energy) criteria
+ *---------------------------------------------------------------*/
+
+size_t WebRtcIlbcfix_XcorrCoef(
+    int16_t *target,  /* (i) first array */
+    int16_t *regressor, /* (i) second array */
+    size_t subl,  /* (i) dimension arrays */
+    size_t searchLen, /* (i) the search lenght */
+    size_t offset,  /* (i) samples offset between arrays */
+    int16_t step   /* (i) +1 or -1 */
+                            ){
+  size_t k;
+  size_t maxlag;
+  int16_t pos;
+  int16_t max;
+  int16_t crossCorrScale, Energyscale;
+  int16_t crossCorrSqMod, crossCorrSqMod_Max;
+  int32_t crossCorr, Energy;
+  int16_t crossCorrmod, EnergyMod, EnergyMod_Max;
+  int16_t *tp, *rp;
+  int16_t *rp_beg, *rp_end;
+  int16_t totscale, totscale_max;
+  int16_t scalediff;
+  int32_t newCrit, maxCrit;
+  int shifts;
+
+  /* Initializations, to make sure that the first one is selected */
+  crossCorrSqMod_Max=0;
+  EnergyMod_Max=WEBRTC_SPL_WORD16_MAX;
+  totscale_max=-500;
+  maxlag=0;
+  pos=0;
+
+  /* Find scale value and start position */
+  if (step==1) {
+    max=WebRtcSpl_MaxAbsValueW16(regressor, subl + searchLen - 1);
+    rp_beg = regressor;
+    rp_end = regressor + subl;
+  } else { /* step==-1 */
+    max = WebRtcSpl_MaxAbsValueW16(regressor - searchLen, subl + searchLen - 1);
+    rp_beg = regressor - 1;
+    rp_end = regressor + subl - 1;
+  }
+
+  /* Introduce a scale factor on the Energy in int32_t in
+     order to make sure that the calculation does not
+     overflow */
+
+  if (max>5000) {
+    shifts=2;
+  } else {
+    shifts=0;
+  }
+
+  /* Calculate the first energy, then do a +/- to get the other energies */
+  Energy=WebRtcSpl_DotProductWithScale(regressor, regressor, subl, shifts);
+
+  for (k=0;k<searchLen;k++) {
+    tp = target;
+    rp = &regressor[pos];
+
+    crossCorr=WebRtcSpl_DotProductWithScale(tp, rp, subl, shifts);
+
+    if ((Energy>0)&&(crossCorr>0)) {
+
+      /* Put cross correlation and energy on 16 bit word */
+      crossCorrScale=(int16_t)WebRtcSpl_NormW32(crossCorr)-16;
+      crossCorrmod=(int16_t)WEBRTC_SPL_SHIFT_W32(crossCorr, crossCorrScale);
+      Energyscale=(int16_t)WebRtcSpl_NormW32(Energy)-16;
+      EnergyMod=(int16_t)WEBRTC_SPL_SHIFT_W32(Energy, Energyscale);
+
+      /* Square cross correlation and store upper int16_t */
+      crossCorrSqMod = (int16_t)((crossCorrmod * crossCorrmod) >> 16);
+
+      /* Calculate the total number of (dynamic) right shifts that have
+         been performed on (crossCorr*crossCorr)/energy
+      */
+      totscale=Energyscale-(crossCorrScale<<1);
+
+      /* Calculate the shift difference in order to be able to compare the two
+         (crossCorr*crossCorr)/energy in the same domain
+      */
+      scalediff=totscale-totscale_max;
+      scalediff=WEBRTC_SPL_MIN(scalediff,31);
+      scalediff=WEBRTC_SPL_MAX(scalediff,-31);
+
+      /* Compute the cross multiplication between the old best criteria
+         and the new one to be able to compare them without using a
+         division */
+
+      if (scalediff<0) {
+        newCrit = ((int32_t)crossCorrSqMod*EnergyMod_Max)>>(-scalediff);
+        maxCrit = ((int32_t)crossCorrSqMod_Max*EnergyMod);
+      } else {
+        newCrit = ((int32_t)crossCorrSqMod*EnergyMod_Max);
+        maxCrit = ((int32_t)crossCorrSqMod_Max*EnergyMod)>>scalediff;
+      }
+
+      /* Store the new lag value if the new criteria is larger
+         than previous largest criteria */
+
+      if (newCrit > maxCrit) {
+        crossCorrSqMod_Max = crossCorrSqMod;
+        EnergyMod_Max = EnergyMod;
+        totscale_max = totscale;
+        maxlag = k;
+      }
+    }
+    pos+=step;
+
+    /* Do a +/- to get the next energy */
+    Energy += step * ((*rp_end * *rp_end - *rp_beg * *rp_beg) >> shifts);
+    rp_beg+=step;
+    rp_end+=step;
+  }
+
+  return(maxlag+offset);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
new file mode 100644
index 0000000000..3be5a296b5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/******************************************************************
+
+ iLBC Speech Coder ANSI-C Source Code
+
+ WebRtcIlbcfix_XcorrCoef.h
+
+******************************************************************/
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+/*----------------------------------------------------------------*
+ * cross correlation which finds the optimal lag for the
+ * crossCorr*crossCorr/(energy) criteria
+ *---------------------------------------------------------------*/
+
+size_t WebRtcIlbcfix_XcorrCoef(
+    int16_t* target,    /* (i) first array */
+    int16_t* regressor, /* (i) second array */
+    size_t subl,        /* (i) dimension arrays */
+    size_t searchLen,   /* (i) the search lenght */
+    size_t offset,      /* (i) samples offset between arrays */
+    int16_t step        /* (i) +1 or -1 */
+    );
+
+#endif
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/bandwidth_info.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/bandwidth_info.h
new file mode 100644
index 0000000000..c3830a5f7c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/bandwidth_info.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+
+#include <stdint.h>
+
+typedef struct {
+  int in_use;
+  int32_t send_bw_avg;
+  int32_t send_max_delay_avg;
+  int16_t bottleneck_idx;
+  int16_t jitter_info;
+} IsacBandwidthInfo;
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
new file mode 100644
index 0000000000..a4f297c5a1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
@@ -0,0 +1,195 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory.h>
+#include <string.h>
+#ifdef WEBRTC_ANDROID
+#include <stdlib.h>
+#endif
+
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
+
+static void WebRtcIsac_AllPoleFilter(double* InOut,
+                                     double* Coef,
+                                     size_t lengthInOut,
+                                     int orderCoef) {
+  /* the state of filter is assumed to be in InOut[-1] to InOut[-orderCoef] */
+  double scal;
+  double sum;
+  size_t n;
+  int k;
+
+  //if (fabs(Coef[0]-1.0)<0.001) {
+  if ( (Coef[0] > 0.9999) && (Coef[0] < 1.0001) )
+  {
+    for(n = 0; n < lengthInOut; n++)
+    {
+      sum = Coef[1] * InOut[-1];
+      for(k = 2; k <= orderCoef; k++){
+        sum += Coef[k] * InOut[-k];
+      }
+      *InOut++ -= sum;
+    }
+  }
+  else
+  {
+    scal = 1.0 / Coef[0];
+    for(n=0;n<lengthInOut;n++)
+    {
+      *InOut *= scal;
+      for(k=1;k<=orderCoef;k++){
+        *InOut -= scal*Coef[k]*InOut[-k];
+      }
+      InOut++;
+    }
+  }
+}
+
+static void WebRtcIsac_AllZeroFilter(double* In,
+                                     double* Coef,
+                                     size_t lengthInOut,
+                                     int orderCoef,
+                                     double* Out) {
+  /* the state of filter is assumed to be in In[-1] to In[-orderCoef] */
+
+  size_t n;
+  int k;
+  double tmp;
+
+  for(n = 0; n < lengthInOut; n++)
+  {
+    tmp = In[0] * Coef[0];
+
+    for(k = 1; k <= orderCoef; k++){
+      tmp += Coef[k] * In[-k];
+    }
+
+    *Out++ = tmp;
+    In++;
+  }
+}
+
+static void WebRtcIsac_ZeroPoleFilter(double* In,
+                                      double* ZeroCoef,
+                                      double* PoleCoef,
+                                      size_t lengthInOut,
+                                      int orderCoef,
+                                      double* Out) {
+  /* the state of the zero section is assumed to be in In[-1] to In[-orderCoef] */
+  /* the state of the pole section is assumed to be in Out[-1] to Out[-orderCoef] */
+
+  WebRtcIsac_AllZeroFilter(In,ZeroCoef,lengthInOut,orderCoef,Out);
+  WebRtcIsac_AllPoleFilter(Out,PoleCoef,lengthInOut,orderCoef);
+}
+
+
+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order) {
+  size_t  lag, n;
+  double sum, prod;
+  const double *x_lag;
+
+  for (lag = 0; lag <= order; lag++)
+  {
+    sum = 0.0f;
+    x_lag = &x[lag];
+    prod = x[0] * x_lag[0];
+    for (n = 1; n < N - lag; n++) {
+      sum += prod;
+      prod = x[n] * x_lag[n];
+    }
+    sum += prod;
+    r[lag] = sum;
+  }
+
+}
+
+static void WebRtcIsac_BwExpand(double* out,
+                                double* in,
+                                double coef,
+                                size_t length) {
+  size_t i;
+  double  chirp;
+
+  chirp = coef;
+
+  out[0] = in[0];
+  for (i = 1; i < length; i++) {
+    out[i] = chirp * in[i];
+    chirp *= coef;
+  }
+}
+
+void WebRtcIsac_WeightingFilter(const double* in,
+                                double* weiout,
+                                double* whiout,
+                                WeightFiltstr* wfdata) {
+  double  tmpbuffer[PITCH_FRAME_LEN + PITCH_WLPCBUFLEN];
+  double  corr[PITCH_WLPCORDER+1], rc[PITCH_WLPCORDER+1];
+  double apol[PITCH_WLPCORDER+1], apolr[PITCH_WLPCORDER+1];
+  double  rho=0.9, *inp, *dp, *dp2;
+  double  whoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
+  double  weoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
+  double  *weo, *who, opol[PITCH_WLPCORDER+1], ext[PITCH_WLPCWINLEN];
+  int     k, n, endpos, start;
+
+  /* Set up buffer and states */
+  memcpy(tmpbuffer, wfdata->buffer, sizeof(double) * PITCH_WLPCBUFLEN);
+  memcpy(tmpbuffer+PITCH_WLPCBUFLEN, in, sizeof(double) * PITCH_FRAME_LEN);
+  memcpy(wfdata->buffer, tmpbuffer+PITCH_FRAME_LEN, sizeof(double) * PITCH_WLPCBUFLEN);
+
+  dp=weoutbuf;
+  dp2=whoutbuf;
+  for (k=0;k<PITCH_WLPCORDER;k++) {
+    *dp++ = wfdata->weostate[k];
+    *dp2++ = wfdata->whostate[k];
+    opol[k]=0.0;
+  }
+  opol[0]=1.0;
+  opol[PITCH_WLPCORDER]=0.0;
+  weo=dp;
+  who=dp2;
+
+  endpos=PITCH_WLPCBUFLEN + PITCH_SUBFRAME_LEN;
+  inp=tmpbuffer + PITCH_WLPCBUFLEN;
+
+  for (n=0; n<PITCH_SUBFRAMES; n++) {
+    /* Windowing */
+    start=endpos-PITCH_WLPCWINLEN;
+    for (k=0; k<PITCH_WLPCWINLEN; k++) {
+      ext[k]=wfdata->window[k]*tmpbuffer[start+k];
+    }
+
+    /* Get LPC polynomial */
+    WebRtcIsac_AutoCorr(corr, ext, PITCH_WLPCWINLEN, PITCH_WLPCORDER);
+    corr[0]=1.01*corr[0]+1.0; /* White noise correction */
+    WebRtcIsac_LevDurb(apol, rc, corr, PITCH_WLPCORDER);
+    WebRtcIsac_BwExpand(apolr, apol, rho, PITCH_WLPCORDER+1);
+
+    /* Filtering */
+    WebRtcIsac_ZeroPoleFilter(inp, apol, apolr, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, weo);
+    WebRtcIsac_ZeroPoleFilter(inp, apolr, opol, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, who);
+
+    inp+=PITCH_SUBFRAME_LEN;
+    endpos+=PITCH_SUBFRAME_LEN;
+    weo+=PITCH_SUBFRAME_LEN;
+    who+=PITCH_SUBFRAME_LEN;
+  }
+
+  /* Export filter states */
+  for (k=0;k<PITCH_WLPCORDER;k++) {
+    wfdata->weostate[k]=weoutbuf[PITCH_FRAME_LEN+k];
+    wfdata->whostate[k]=whoutbuf[PITCH_FRAME_LEN+k];
+  }
+
+  /* Export output data */
+  memcpy(weiout, weoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
+  memcpy(whiout, whoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.h
new file mode 100644
index 0000000000..a747a7f549
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
+
+#include <stddef.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
+
+void WebRtcIsac_WeightingFilter(const double* in,
+                                double* weiout,
+                                double* whiout,
+                                WeightFiltstr* wfdata);
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c
new file mode 100644
index 0000000000..57cf0c39da
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c
@@ -0,0 +1,409 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
+
+#include <math.h>
+
+void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata) {
+  int k;
+
+  for (k = 0; k < PITCH_BUFFSIZE; k++) {
+    pitchfiltdata->ubuf[k] = 0.0;
+  }
+  pitchfiltdata->ystate[0] = 0.0;
+  for (k = 1; k < (PITCH_DAMPORDER); k++) {
+    pitchfiltdata->ystate[k] = 0.0;
+  }
+  pitchfiltdata->oldlagp[0] = 50.0;
+  pitchfiltdata->oldgainp[0] = 0.0;
+}
+
+static void WebRtcIsac_InitWeightingFilter(WeightFiltstr* wfdata) {
+  int k;
+  double t, dtmp, dtmp2, denum, denum2;
+
+  for (k = 0; k < PITCH_WLPCBUFLEN; k++)
+    wfdata->buffer[k] = 0.0;
+
+  for (k = 0; k < PITCH_WLPCORDER; k++) {
+    wfdata->istate[k] = 0.0;
+    wfdata->weostate[k] = 0.0;
+    wfdata->whostate[k] = 0.0;
+  }
+
+  /* next part should be in Matlab, writing to a global table */
+  t = 0.5;
+  denum = 1.0 / ((double)PITCH_WLPCWINLEN);
+  denum2 = denum * denum;
+  for (k = 0; k < PITCH_WLPCWINLEN; k++) {
+    dtmp = PITCH_WLPCASYM * t * denum + (1 - PITCH_WLPCASYM) * t * t * denum2;
+    dtmp *= 3.14159265;
+    dtmp2 = sin(dtmp);
+    wfdata->window[k] = dtmp2 * dtmp2;
+    t++;
+  }
+}
+
+void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* State) {
+  int k;
+
+  for (k = 0; k < PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
+                      PITCH_FRAME_LEN / 2 + 2;
+       k++)
+    State->dec_buffer[k] = 0.0;
+  for (k = 0; k < 2 * ALLPASSSECTIONS + 1; k++)
+    State->decimator_state[k] = 0.0;
+  for (k = 0; k < 2; k++)
+    State->hp_state[k] = 0.0;
+  for (k = 0; k < QLOOKAHEAD; k++)
+    State->whitened_buf[k] = 0.0;
+  for (k = 0; k < QLOOKAHEAD; k++)
+    State->inbuf[k] = 0.0;
+
+  WebRtcIsac_InitPitchFilter(&(State->PFstr_wght));
+
+  WebRtcIsac_InitPitchFilter(&(State->PFstr));
+
+  WebRtcIsac_InitWeightingFilter(&(State->Wghtstr));
+}
+
+void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata) {
+  int k;
+
+  for (k = 0; k < QLOOKAHEAD; k++) {
+    prefiltdata->INLABUF1[k] = 0;
+    prefiltdata->INLABUF2[k] = 0;
+
+    prefiltdata->INLABUF1_float[k] = 0;
+    prefiltdata->INLABUF2_float[k] = 0;
+  }
+  for (k = 0; k < 2 * (QORDER - 1); k++) {
+    prefiltdata->INSTAT1[k] = 0;
+    prefiltdata->INSTAT2[k] = 0;
+    prefiltdata->INSTATLA1[k] = 0;
+    prefiltdata->INSTATLA2[k] = 0;
+
+    prefiltdata->INSTAT1_float[k] = 0;
+    prefiltdata->INSTAT2_float[k] = 0;
+    prefiltdata->INSTATLA1_float[k] = 0;
+    prefiltdata->INSTATLA2_float[k] = 0;
+  }
+
+  /* High pass filter states */
+  prefiltdata->HPstates[0] = 0.0;
+  prefiltdata->HPstates[1] = 0.0;
+
+  prefiltdata->HPstates_float[0] = 0.0f;
+  prefiltdata->HPstates_float[1] = 0.0f;
+
+  return;
+}
+
+double WebRtcIsac_LevDurb(double* a, double* k, double* r, size_t order) {
+  const double LEVINSON_EPS = 1.0e-10;
+
+  double sum, alpha;
+  size_t m, m_h, i;
+  alpha = 0;  // warning -DH
+  a[0] = 1.0;
+  if (r[0] < LEVINSON_EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
+    for (i = 0; i < order; i++) {
+      k[i] = 0;
+      a[i + 1] = 0;
+    }
+  } else {
+    a[1] = k[0] = -r[1] / r[0];
+    alpha = r[0] + r[1] * k[0];
+    for (m = 1; m < order; m++) {
+      sum = r[m + 1];
+      for (i = 0; i < m; i++) {
+        sum += a[i + 1] * r[m - i];
+      }
+      k[m] = -sum / alpha;
+      alpha += k[m] * sum;
+      m_h = (m + 1) >> 1;
+      for (i = 0; i < m_h; i++) {
+        sum = a[i + 1] + k[m] * a[m - i];
+        a[m - i] += k[m] * a[i + 1];
+        a[i + 1] = sum;
+      }
+      a[m + 1] = k[m];
+    }
+  }
+  return alpha;
+}
+
+/* The upper channel all-pass filter factors */
+const float WebRtcIsac_kUpperApFactorsFloat[2] = {0.03470000000000f,
+                                                  0.38260000000000f};
+
+/* The lower channel all-pass filter factors */
+const float WebRtcIsac_kLowerApFactorsFloat[2] = {0.15440000000000f,
+                                                  0.74400000000000f};
+
+/* This function performs all-pass filtering--a series of first order all-pass
+ * sections are used to filter the input in a cascade manner.
+ * The input is overwritten!!
+ */
+void WebRtcIsac_AllPassFilter2Float(float* InOut,
+                                    const float* APSectionFactors,
+                                    int lengthInOut,
+                                    int NumberOfSections,
+                                    float* FilterState) {
+  int n, j;
+  float temp;
+  for (j = 0; j < NumberOfSections; j++) {
+    for (n = 0; n < lengthInOut; n++) {
+      temp = FilterState[j] + APSectionFactors[j] * InOut[n];
+      FilterState[j] = -APSectionFactors[j] * temp + InOut[n];
+      InOut[n] = temp;
+    }
+  }
+}
+
+/* The number of composite all-pass filter factors */
+#define NUMBEROFCOMPOSITEAPSECTIONS 4
+
+/* Function WebRtcIsac_SplitAndFilter
+ * This function creates low-pass and high-pass decimated versions of part of
+ the input signal, and part of the signal in the input 'lookahead buffer'.
+
+ INPUTS:
+ in: a length FRAMESAMPLES array of input samples
+ prefiltdata: input data structure containing the filterbank states
+ and lookahead samples from the previous encoding
+ iteration.
+ OUTPUTS:
+ LP: a FRAMESAMPLES_HALF array of low-pass filtered samples that
+ have been phase equalized.  The first QLOOKAHEAD samples are
+ based on the samples in the two prefiltdata->INLABUFx arrays
+ each of length QLOOKAHEAD.
+ The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
+ on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
+ array in[].
+ HP: a FRAMESAMPLES_HALF array of high-pass filtered samples that
+ have been phase equalized.  The first QLOOKAHEAD samples are
+ based on the samples in the two prefiltdata->INLABUFx arrays
+ each of length QLOOKAHEAD.
+ The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
+ on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
+ array in[].
+
+ LP_la: a FRAMESAMPLES_HALF array of low-pass filtered samples.
+ These samples are not phase equalized. They are computed
+ from the samples in the in[] array.
+ HP_la: a FRAMESAMPLES_HALF array of high-pass filtered samples
+ that are not phase equalized. They are computed from
+ the in[] vector.
+ prefiltdata: this input data structure's filterbank state and
+ lookahead sample buffers are updated for the next
+ encoding iteration.
+*/
+void WebRtcIsac_SplitAndFilterFloat(float* pin,
+                                    float* LP,
+                                    float* HP,
+                                    double* LP_la,
+                                    double* HP_la,
+                                    PreFiltBankstr* prefiltdata) {
+  int k, n;
+  float CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
+  float ForTransform_CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
+  float ForTransform_CompositeAPFilterState2[NUMBEROFCOMPOSITEAPSECTIONS];
+  float tempinoutvec[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
+  float tempin_ch1[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
+  float tempin_ch2[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
+  float in[FRAMESAMPLES];
+  float ftmp;
+
+  /* HPstcoeff_in = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
+  static const float kHpStCoefInFloat[4] = {
+      -1.94895953203325f, 0.94984516000000f, -0.05101826139794f,
+      0.05015484000000f};
+
+  /* The composite all-pass filter factors */
+  static const float WebRtcIsac_kCompositeApFactorsFloat[4] = {
+      0.03470000000000f, 0.15440000000000f, 0.38260000000000f,
+      0.74400000000000f};
+
+  // The matrix for transforming the backward composite state to upper channel
+  // state.
+  static const float WebRtcIsac_kTransform1Float[8] = {
+      -0.00158678506084f, 0.00127157815343f, -0.00104805672709f,
+      0.00084837248079f,  0.00134467983258f, -0.00107756549387f,
+      0.00088814793277f,  -0.00071893072525f};
+
+  // The matrix for transforming the backward composite state to lower channel
+  // state.
+  static const float WebRtcIsac_kTransform2Float[8] = {
+      -0.00170686041697f, 0.00136780109829f, -0.00112736532350f,
+      0.00091257055385f,  0.00103094281812f, -0.00082615076557f,
+      0.00068092756088f,  -0.00055119165484f};
+
+  /* High pass filter */
+
+  for (k = 0; k < FRAMESAMPLES; k++) {
+    in[k] = pin[k] + kHpStCoefInFloat[2] * prefiltdata->HPstates_float[0] +
+            kHpStCoefInFloat[3] * prefiltdata->HPstates_float[1];
+    ftmp = pin[k] - kHpStCoefInFloat[0] * prefiltdata->HPstates_float[0] -
+           kHpStCoefInFloat[1] * prefiltdata->HPstates_float[1];
+    prefiltdata->HPstates_float[1] = prefiltdata->HPstates_float[0];
+    prefiltdata->HPstates_float[0] = ftmp;
+  }
+
+  /* First Channel */
+
+  /*initial state of composite filter is zero */
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    CompositeAPFilterState[k] = 0.0;
+  }
+  /* put every other sample of input into a temporary vector in reverse
+   * (backward) order*/
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempinoutvec[k] = in[FRAMESAMPLES - 1 - 2 * k];
+  }
+
+  /* now all-pass filter the backwards vector.  Output values overwrite the
+   * input vector. */
+  WebRtcIsac_AllPassFilter2Float(
+      tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat, FRAMESAMPLES_HALF,
+      NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  /* save the backwards filtered output for later forward filtering,
+     but write it in forward order*/
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempin_ch1[FRAMESAMPLES_HALF + QLOOKAHEAD - 1 - k] = tempinoutvec[k];
+  }
+
+  /* save the backwards filter state  becaue it will be transformed
+     later into a forward state */
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    ForTransform_CompositeAPFilterState[k] = CompositeAPFilterState[k];
+  }
+
+  /* now backwards filter the samples in the lookahead buffer. The samples were
+     placed there in the encoding of the previous frame.  The output samples
+     overwrite the input samples */
+  WebRtcIsac_AllPassFilter2Float(
+      prefiltdata->INLABUF1_float, WebRtcIsac_kCompositeApFactorsFloat,
+      QLOOKAHEAD, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  /* save the output, but write it in forward order */
+  /* write the lookahead samples for the next encoding iteration. Every other
+     sample at the end of the input frame is written in reverse order for the
+     lookahead length. Exported in the prefiltdata structure. */
+  for (k = 0; k < QLOOKAHEAD; k++) {
+    tempin_ch1[QLOOKAHEAD - 1 - k] = prefiltdata->INLABUF1_float[k];
+    prefiltdata->INLABUF1_float[k] = in[FRAMESAMPLES - 1 - 2 * k];
+  }
+
+  /* Second Channel.  This is exactly like the first channel, except that the
+     even samples are now filtered instead (lower channel). */
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    CompositeAPFilterState[k] = 0.0;
+  }
+
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempinoutvec[k] = in[FRAMESAMPLES - 2 - 2 * k];
+  }
+
+  WebRtcIsac_AllPassFilter2Float(
+      tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat, FRAMESAMPLES_HALF,
+      NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempin_ch2[FRAMESAMPLES_HALF + QLOOKAHEAD - 1 - k] = tempinoutvec[k];
+  }
+
+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
+    ForTransform_CompositeAPFilterState2[k] = CompositeAPFilterState[k];
+  }
+
+  WebRtcIsac_AllPassFilter2Float(
+      prefiltdata->INLABUF2_float, WebRtcIsac_kCompositeApFactorsFloat,
+      QLOOKAHEAD, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
+
+  for (k = 0; k < QLOOKAHEAD; k++) {
+    tempin_ch2[QLOOKAHEAD - 1 - k] = prefiltdata->INLABUF2_float[k];
+    prefiltdata->INLABUF2_float[k] = in[FRAMESAMPLES - 2 - 2 * k];
+  }
+
+  /* Transform filter states from backward to forward */
+  /*At this point, each of the states of the backwards composite filters for the
+    two channels are transformed into forward filtering states for the
+    corresponding forward channel filters.  Each channel's forward filtering
+    state from the previous
+    encoding iteration is added to the transformed state to get a proper forward
+    state */
+
+  /* So the existing NUMBEROFCOMPOSITEAPSECTIONS x 1 (4x1) state vector is
+     multiplied by a NUMBEROFCHANNELAPSECTIONSxNUMBEROFCOMPOSITEAPSECTIONS (2x4)
+     transform matrix to get the new state that is added to the previous 2x1
+     input state */
+
+  for (k = 0; k < NUMBEROFCHANNELAPSECTIONS; k++) { /* k is row variable */
+    for (n = 0; n < NUMBEROFCOMPOSITEAPSECTIONS;
+         n++) { /* n is column variable */
+      prefiltdata->INSTAT1_float[k] +=
+          ForTransform_CompositeAPFilterState[n] *
+          WebRtcIsac_kTransform1Float[k * NUMBEROFCHANNELAPSECTIONS + n];
+      prefiltdata->INSTAT2_float[k] +=
+          ForTransform_CompositeAPFilterState2[n] *
+          WebRtcIsac_kTransform2Float[k * NUMBEROFCHANNELAPSECTIONS + n];
+    }
+  }
+
+  /*obtain polyphase components by forward all-pass filtering through each
+   * channel */
+  /* the backward filtered samples are now forward filtered with the
+   * corresponding channel filters */
+  /* The all pass filtering automatically updates the filter states which are
+     exported in the prefiltdata structure */
+  WebRtcIsac_AllPassFilter2Float(tempin_ch1, WebRtcIsac_kUpperApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTAT1_float);
+  WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kLowerApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTAT2_float);
+
+  /* Now Construct low-pass and high-pass signals as combinations of polyphase
+   * components */
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    LP[k] = 0.5f * (tempin_ch1[k] + tempin_ch2[k]); /* low pass signal*/
+    HP[k] = 0.5f * (tempin_ch1[k] - tempin_ch2[k]); /* high pass signal*/
+  }
+
+  /* Lookahead LP and HP signals */
+  /* now create low pass and high pass signals of the input vector.  However, no
+     backwards filtering is performed, and hence no phase equalization is
+     involved. Also, the input contains some samples that are lookahead samples.
+     The high pass and low pass signals that are created are used outside this
+     function for analysis (not encoding) purposes */
+
+  /* set up input */
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    tempin_ch1[k] = in[2 * k + 1];
+    tempin_ch2[k] = in[2 * k];
+  }
+
+  /* the input filter states are passed in and updated by the all-pass filtering
+     routine and exported in the prefiltdata structure*/
+  WebRtcIsac_AllPassFilter2Float(tempin_ch1, WebRtcIsac_kUpperApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTATLA1_float);
+  WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kLowerApFactorsFloat,
+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
+                                 prefiltdata->INSTATLA2_float);
+
+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
+    LP_la[k] = (float)(0.5f * (tempin_ch1[k] + tempin_ch2[k]));  /*low pass */
+    HP_la[k] = (double)(0.5f * (tempin_ch1[k] - tempin_ch2[k])); /* high pass */
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.h
new file mode 100644
index 0000000000..1aecfc4046
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
+
+#include <stddef.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata);
+void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* state);
+void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata);
+
+double WebRtcIsac_LevDurb(double* a, double* k, double* r, size_t order);
+
+/* The number of all-pass filter factors in an upper or lower channel*/
+#define NUMBEROFCHANNELAPSECTIONS 2
+
+/* The upper channel all-pass filter factors */
+extern const float WebRtcIsac_kUpperApFactorsFloat[2];
+
+/* The lower channel all-pass filter factors */
+extern const float WebRtcIsac_kLowerApFactorsFloat[2];
+
+void WebRtcIsac_AllPassFilter2Float(float* InOut,
+                                    const float* APSectionFactors,
+                                    int lengthInOut,
+                                    int NumberOfSections,
+                                    float* FilterState);
+void WebRtcIsac_SplitAndFilterFloat(float* in,
+                                    float* LP,
+                                    float* HP,
+                                    double* LP_la,
+                                    double* HP_la,
+                                    PreFiltBankstr* prefiltdata);
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
new file mode 100644
index 0000000000..fe9afa4ba2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+
+#include <math.h>
+
+#include "rtc_base/system/arch.h"
+
+#if defined(WEBRTC_POSIX)
+#define WebRtcIsac_lrint lrint
+#elif (defined(WEBRTC_ARCH_X86) && defined(WIN32))
+static __inline long int WebRtcIsac_lrint(double x_dbl) {
+  long int x_int;
+
+  __asm {
+    fld x_dbl
+    fistp x_int
+  }
+  ;
+
+  return x_int;
+}
+#else  // Do a slow but correct implementation of lrint
+
+static __inline long int WebRtcIsac_lrint(double x_dbl) {
+  long int x_int;
+  x_int = (long int)floor(x_dbl + 0.499999999999);
+  return x_int;
+}
+
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
new file mode 100644
index 0000000000..8a19ac1710
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
@@ -0,0 +1,695 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+
+#include <math.h>
+#include <memory.h>
+#include <string.h>
+#ifdef WEBRTC_ANDROID
+#include <stdlib.h>
+#endif
+
+#include "modules/audio_coding/codecs/isac/main/source/filter_functions.h"
+#include "modules/audio_coding/codecs/isac/main/source/pitch_filter.h"
+#include "rtc_base/system/ignore_warnings.h"
+
+static const double kInterpolWin[8] = {-0.00067556028640,  0.02184247643159, -0.12203175715679,  0.60086484101160,
+                                       0.60086484101160, -0.12203175715679,  0.02184247643159, -0.00067556028640};
+
+/* interpolation filter */
+__inline static void IntrepolFilter(double *data_ptr, double *intrp)
+{
+  *intrp = kInterpolWin[0] * data_ptr[-3];
+  *intrp += kInterpolWin[1] * data_ptr[-2];
+  *intrp += kInterpolWin[2] * data_ptr[-1];
+  *intrp += kInterpolWin[3] * data_ptr[0];
+  *intrp += kInterpolWin[4] * data_ptr[1];
+  *intrp += kInterpolWin[5] * data_ptr[2];
+  *intrp += kInterpolWin[6] * data_ptr[3];
+  *intrp += kInterpolWin[7] * data_ptr[4];
+}
+
+
+/* 2D parabolic interpolation */
+/* probably some 0.5 factors can be eliminated, and the square-roots can be removed from the Cholesky fact. */
+__inline static void Intrpol2D(double T[3][3], double *x, double *y, double *peak_val)
+{
+  double c, b[2], A[2][2];
+  double t1, t2, d;
+  double delta1, delta2;
+
+
+  // double T[3][3] = {{-1.25, -.25,-.25}, {-.25, .75, .75}, {-.25, .75, .75}};
+  // should result in: delta1 = 0.5;  delta2 = 0.0;  peak_val = 1.0
+
+  c = T[1][1];
+  b[0] = 0.5 * (T[1][2] + T[2][1] - T[0][1] - T[1][0]);
+  b[1] = 0.5 * (T[1][0] + T[2][1] - T[0][1] - T[1][2]);
+  A[0][1] = -0.5 * (T[0][1] + T[2][1] - T[1][0] - T[1][2]);
+  t1 = 0.5 * (T[0][0] + T[2][2]) - c;
+  t2 = 0.5 * (T[2][0] + T[0][2]) - c;
+  d = (T[0][1] + T[1][2] + T[1][0] + T[2][1]) - 4.0 * c - t1 - t2;
+  A[0][0] = -t1 - 0.5 * d;
+  A[1][1] = -t2 - 0.5 * d;
+
+  /* deal with singularities or ill-conditioned cases */
+  if ( (A[0][0] < 1e-7) || ((A[0][0] * A[1][1] - A[0][1] * A[0][1]) < 1e-7) ) {
+    *peak_val = T[1][1];
+    return;
+  }
+
+  /* Cholesky decomposition: replace A by upper-triangular factor */
+  A[0][0] = sqrt(A[0][0]);
+  A[0][1] = A[0][1] / A[0][0];
+  A[1][1] = sqrt(A[1][1] - A[0][1] * A[0][1]);
+
+  /* compute [x; y] = -0.5 * inv(A) * b */
+  t1 = b[0] / A[0][0];
+  t2 = (b[1] - t1 * A[0][1]) / A[1][1];
+  delta2 = t2 / A[1][1];
+  delta1 = 0.5 * (t1 - delta2 * A[0][1]) / A[0][0];
+  delta2 *= 0.5;
+
+  /* limit norm */
+  t1 = delta1 * delta1 + delta2 * delta2;
+  if (t1 > 1.0) {
+    delta1 /= t1;
+    delta2 /= t1;
+  }
+
+  *peak_val = 0.5 * (b[0] * delta1 + b[1] * delta2) + c;
+
+  *x += delta1;
+  *y += delta2;
+}
+
+
+static void PCorr(const double *in, double *outcorr)
+{
+  double sum, ysum, prod;
+  const double *x, *inptr;
+  int k, n;
+
+  //ysum = 1e-6;          /* use this with float (i.s.o. double)! */
+  ysum = 1e-13;
+  sum = 0.0;
+  x = in + PITCH_MAX_LAG/2 + 2;
+  for (n = 0; n < PITCH_CORR_LEN2; n++) {
+    ysum += in[n] * in[n];
+    sum += x[n] * in[n];
+  }
+
+  outcorr += PITCH_LAG_SPAN2 - 1;     /* index of last element in array */
+  *outcorr = sum / sqrt(ysum);
+
+  for (k = 1; k < PITCH_LAG_SPAN2; k++) {
+    ysum -= in[k-1] * in[k-1];
+    ysum += in[PITCH_CORR_LEN2 + k - 1] * in[PITCH_CORR_LEN2 + k - 1];
+    sum = 0.0;
+    inptr = &in[k];
+    prod = x[0] * inptr[0];
+    for (n = 1; n < PITCH_CORR_LEN2; n++) {
+      sum += prod;
+      prod = x[n] * inptr[n];
+    }
+    sum += prod;
+    outcorr--;
+    *outcorr = sum / sqrt(ysum);
+  }
+}
+
+static void WebRtcIsac_AllpassFilterForDec(double* InOut,
+                                           const double* APSectionFactors,
+                                           size_t lengthInOut,
+                                           double* FilterState) {
+  // This performs all-pass filtering--a series of first order all-pass
+  // sections are used to filter the input in a cascade manner.
+  size_t n, j;
+  double temp;
+  for (j = 0; j < ALLPASSSECTIONS; j++) {
+    for (n = 0; n < lengthInOut; n += 2) {
+      temp = InOut[n];  // store input
+      InOut[n] = FilterState[j] + APSectionFactors[j] * temp;
+      FilterState[j] = -APSectionFactors[j] * InOut[n] + temp;
+    }
+  }
+}
+
+static void WebRtcIsac_DecimateAllpass(
+    const double* in,
+    double* state_in,  // array of size: 2*ALLPASSSECTIONS+1
+    size_t N,          // number of input samples
+    double* out) {     // array of size N/2
+
+  static const double APupper[ALLPASSSECTIONS] = {0.0347, 0.3826};
+  static const double APlower[ALLPASSSECTIONS] = {0.1544, 0.744};
+
+  size_t n;
+  double data_vec[PITCH_FRAME_LEN];
+
+  /* copy input */
+  memcpy(data_vec + 1, in, sizeof(double) * (N - 1));
+
+  data_vec[0] = state_in[2 * ALLPASSSECTIONS];  // the z^(-1) state
+  state_in[2 * ALLPASSSECTIONS] = in[N - 1];
+
+  WebRtcIsac_AllpassFilterForDec(data_vec + 1, APupper, N, state_in);
+  WebRtcIsac_AllpassFilterForDec(data_vec, APlower, N,
+                                 state_in + ALLPASSSECTIONS);
+
+  for (n = 0; n < N / 2; n++)
+    out[n] = data_vec[2 * n] + data_vec[2 * n + 1];
+}
+
+RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
+
+static void WebRtcIsac_InitializePitch(const double* in,
+                                       const double old_lag,
+                                       const double old_gain,
+                                       PitchAnalysisStruct* State,
+                                       double* lags) {
+  double buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
+  double ratio, log_lag, gain_bias;
+  double bias;
+  double corrvec1[PITCH_LAG_SPAN2];
+  double corrvec2[PITCH_LAG_SPAN2];
+  int m, k;
+  // Allocating 10 extra entries at the begining of the CorrSurf
+  double corrSurfBuff[10 + (2*PITCH_BW+3)*(PITCH_LAG_SPAN2+4)];
+  double* CorrSurf[2*PITCH_BW+3];
+  double *CorrSurfPtr1, *CorrSurfPtr2;
+  double LagWin[3] = {0.2, 0.5, 0.98};
+  int ind1, ind2, peaks_ind, peak, max_ind;
+  int peaks[PITCH_MAX_NUM_PEAKS];
+  double adj, gain_tmp;
+  double corr, corr_max;
+  double intrp_a, intrp_b, intrp_c, intrp_d;
+  double peak_vals[PITCH_MAX_NUM_PEAKS];
+  double lags1[PITCH_MAX_NUM_PEAKS];
+  double lags2[PITCH_MAX_NUM_PEAKS];
+  double T[3][3];
+  int row;
+
+  for(k = 0; k < 2*PITCH_BW+3; k++)
+  {
+    CorrSurf[k] = &corrSurfBuff[10 + k * (PITCH_LAG_SPAN2+4)];
+  }
+  /* reset CorrSurf matrix */
+  memset(corrSurfBuff, 0, sizeof(double) * (10 + (2*PITCH_BW+3) * (PITCH_LAG_SPAN2+4)));
+
+  //warnings -DH
+  max_ind = 0;
+  peak = 0;
+
+  /* copy old values from state buffer */
+  memcpy(buf_dec, State->dec_buffer, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
+
+  /* decimation; put result after the old values */
+  WebRtcIsac_DecimateAllpass(in, State->decimator_state, PITCH_FRAME_LEN,
+                             &buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2]);
+
+  /* low-pass filtering */
+  for (k = PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2; k++)
+    buf_dec[k] += 0.75 * buf_dec[k-1] - 0.25 * buf_dec[k-2];
+
+  /* copy end part back into state buffer */
+  memcpy(State->dec_buffer, buf_dec+PITCH_FRAME_LEN/2, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
+
+  /* compute correlation for first and second half of the frame */
+  PCorr(buf_dec, corrvec1);
+  PCorr(buf_dec + PITCH_CORR_STEP2, corrvec2);
+
+  /* bias towards pitch lag of previous frame */
+  log_lag = log(0.5 * old_lag);
+  gain_bias = 4.0 * old_gain * old_gain;
+  if (gain_bias > 0.8) gain_bias = 0.8;
+  for (k = 0; k < PITCH_LAG_SPAN2; k++)
+  {
+    ratio = log((double) (k + (PITCH_MIN_LAG/2-2))) - log_lag;
+    bias = 1.0 + gain_bias * exp(-5.0 * ratio * ratio);
+    corrvec1[k] *= bias;
+  }
+
+  /* taper correlation functions */
+  for (k = 0; k < 3; k++) {
+    gain_tmp = LagWin[k];
+    corrvec1[k] *= gain_tmp;
+    corrvec2[k] *= gain_tmp;
+    corrvec1[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
+    corrvec2[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
+  }
+
+  corr_max = 0.0;
+  /* fill middle row of correlation surface */
+  ind1 = 0;
+  ind2 = 0;
+  CorrSurfPtr1 = &CorrSurf[PITCH_BW][2];
+  for (k = 0; k < PITCH_LAG_SPAN2; k++) {
+    corr = corrvec1[ind1++] + corrvec2[ind2++];
+    CorrSurfPtr1[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+    }
+  }
+  /* fill first and last rows of correlation surface */
+  ind1 = 0;
+  ind2 = PITCH_BW;
+  CorrSurfPtr1 = &CorrSurf[0][2];
+  CorrSurfPtr2 = &CorrSurf[2*PITCH_BW][PITCH_BW+2];
+  for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW; k++) {
+    ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+    adj = 0.2 * ratio * (2.0 - ratio);   /* adjustment factor; inverse parabola as a function of ratio */
+    corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+    CorrSurfPtr1[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+    }
+    corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+    CorrSurfPtr2[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+    }
+  }
+  /* fill second and next to last rows of correlation surface */
+  ind1 = 0;
+  ind2 = PITCH_BW-1;
+  CorrSurfPtr1 = &CorrSurf[1][2];
+  CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-1][PITCH_BW+1];
+  for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+1; k++) {
+    ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+    adj = 0.9 * ratio * (2.0 - ratio);   /* adjustment factor; inverse parabola as a function of ratio */
+    corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+    CorrSurfPtr1[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+    }
+    corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+    CorrSurfPtr2[k] = corr;
+    if (corr > corr_max) {
+      corr_max = corr;  /* update maximum */
+      max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+    }
+  }
+  /* fill remainder of correlation surface */
+  for (m = 2; m < PITCH_BW; m++) {
+    ind1 = 0;
+    ind2 = PITCH_BW - m;         /* always larger than ind1 */
+    CorrSurfPtr1 = &CorrSurf[m][2];
+    CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-m][PITCH_BW+2-m];
+    for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+m; k++) {
+      ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
+      adj = ratio * (2.0 - ratio);    /* adjustment factor; inverse parabola as a function of ratio */
+      corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
+      CorrSurfPtr1[k] = corr;
+      if (corr > corr_max) {
+        corr_max = corr;  /* update maximum */
+        max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+      }
+      corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
+      CorrSurfPtr2[k] = corr;
+      if (corr > corr_max) {
+        corr_max = corr;  /* update maximum */
+        max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
+      }
+    }
+  }
+
+  /* threshold value to qualify as a peak */
+  corr_max *= 0.6;
+
+  peaks_ind = 0;
+  /* find peaks */
+  for (m = 1; m < PITCH_BW+1; m++) {
+    if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+    CorrSurfPtr1 = &CorrSurf[m][2];
+    for (k = 2; k < PITCH_LAG_SPAN2-PITCH_BW-2+m; k++) {
+      corr = CorrSurfPtr1[k];
+      if (corr > corr_max) {
+        if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
+          if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
+            /* found a peak; store index into matrix */
+            peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+            if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+          }
+        }
+      }
+    }
+  }
+  for (m = PITCH_BW+1; m < 2*PITCH_BW; m++) {
+    if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+    CorrSurfPtr1 = &CorrSurf[m][2];
+    for (k = 2+m-PITCH_BW; k < PITCH_LAG_SPAN2-2; k++) {
+      corr = CorrSurfPtr1[k];
+      if (corr > corr_max) {
+        if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
+          if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
+            /* found a peak; store index into matrix */
+            peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
+            if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
+          }
+        }
+      }
+    }
+  }
+
+  if (peaks_ind > 0) {
+    /* examine each peak */
+    CorrSurfPtr1 = &CorrSurf[0][0];
+    for (k = 0; k < peaks_ind; k++) {
+      peak = peaks[k];
+
+      /* compute four interpolated values around current peak */
+      IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)], &intrp_a);
+      IntrepolFilter(&CorrSurfPtr1[peak - 1            ], &intrp_b);
+      IntrepolFilter(&CorrSurfPtr1[peak                ], &intrp_c);
+      IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)], &intrp_d);
+
+      /* determine maximum of the interpolated values */
+      corr = CorrSurfPtr1[peak];
+      corr_max = intrp_a;
+      if (intrp_b > corr_max) corr_max = intrp_b;
+      if (intrp_c > corr_max) corr_max = intrp_c;
+      if (intrp_d > corr_max) corr_max = intrp_d;
+
+      /* determine where the peak sits and fill a 3x3 matrix around it */
+      row = peak / (PITCH_LAG_SPAN2+4);
+      lags1[k] = (double) ((peak - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
+      lags2[k] = (double) (lags1[k] + PITCH_BW - row);
+      if ( corr > corr_max ) {
+        T[0][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+        T[2][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+        T[1][1] = corr;
+        T[0][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+        T[2][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+        T[1][0] = intrp_a;
+        T[0][1] = intrp_b;
+        T[2][1] = intrp_c;
+        T[1][2] = intrp_d;
+      } else {
+        if (intrp_a == corr_max) {
+          lags1[k] -= 0.5;
+          lags2[k] += 0.5;
+          IntrepolFilter(&CorrSurfPtr1[peak - 2*(PITCH_LAG_SPAN2+5)], &T[0][0]);
+          IntrepolFilter(&CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)], &T[2][0]);
+          T[1][1] = intrp_a;
+          T[0][2] = intrp_b;
+          T[2][2] = intrp_c;
+          T[1][0] = CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)];
+          T[0][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+          T[2][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+          T[1][2] = corr;
+        } else if (intrp_b == corr_max) {
+          lags1[k] -= 0.5;
+          lags2[k] -= 0.5;
+          IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+6)], &T[0][0]);
+          T[2][0] = intrp_a;
+          T[1][1] = intrp_b;
+          IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+3)], &T[0][2]);
+          T[2][2] = intrp_d;
+          T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
+          T[0][1] = CorrSurfPtr1[peak - 1];
+          T[2][1] = corr;
+          T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+        } else if (intrp_c == corr_max) {
+          lags1[k] += 0.5;
+          lags2[k] += 0.5;
+          T[0][0] = intrp_a;
+          IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)], &T[2][0]);
+          T[1][1] = intrp_c;
+          T[0][2] = intrp_d;
+          IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)], &T[2][2]);
+          T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
+          T[0][1] = corr;
+          T[2][1] = CorrSurfPtr1[peak + 1];
+          T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+        } else {
+          lags1[k] += 0.5;
+          lags2[k] -= 0.5;
+          T[0][0] = intrp_b;
+          T[2][0] = intrp_c;
+          T[1][1] = intrp_d;
+          IntrepolFilter(&CorrSurfPtr1[peak + 2*(PITCH_LAG_SPAN2+4)], &T[0][2]);
+          IntrepolFilter(&CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)], &T[2][2]);
+          T[1][0] = corr;
+          T[0][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
+          T[2][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
+          T[1][2] = CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)];
+        }
+      }
+
+      /* 2D parabolic interpolation gives more accurate lags and peak value */
+      Intrpol2D(T, &lags1[k], &lags2[k], &peak_vals[k]);
+    }
+
+    /* determine the highest peak, after applying a bias towards short lags */
+    corr_max = 0.0;
+    for (k = 0; k < peaks_ind; k++) {
+      corr = peak_vals[k] * pow(PITCH_PEAK_DECAY, log(lags1[k] + lags2[k]));
+      if (corr > corr_max) {
+        corr_max = corr;
+        peak = k;
+      }
+    }
+
+    lags1[peak] *= 2.0;
+    lags2[peak] *= 2.0;
+
+    if (lags1[peak] < (double) PITCH_MIN_LAG) lags1[peak] = (double) PITCH_MIN_LAG;
+    if (lags2[peak] < (double) PITCH_MIN_LAG) lags2[peak] = (double) PITCH_MIN_LAG;
+    if (lags1[peak] > (double) PITCH_MAX_LAG) lags1[peak] = (double) PITCH_MAX_LAG;
+    if (lags2[peak] > (double) PITCH_MAX_LAG) lags2[peak] = (double) PITCH_MAX_LAG;
+
+    /* store lags of highest peak in output array */
+    lags[0] = lags1[peak];
+    lags[1] = lags1[peak];
+    lags[2] = lags2[peak];
+    lags[3] = lags2[peak];
+  }
+  else
+  {
+    row = max_ind / (PITCH_LAG_SPAN2+4);
+    lags1[0] = (double) ((max_ind - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
+    lags2[0] = (double) (lags1[0] + PITCH_BW - row);
+
+    if (lags1[0] < (double) PITCH_MIN_LAG) lags1[0] = (double) PITCH_MIN_LAG;
+    if (lags2[0] < (double) PITCH_MIN_LAG) lags2[0] = (double) PITCH_MIN_LAG;
+    if (lags1[0] > (double) PITCH_MAX_LAG) lags1[0] = (double) PITCH_MAX_LAG;
+    if (lags2[0] > (double) PITCH_MAX_LAG) lags2[0] = (double) PITCH_MAX_LAG;
+
+    /* store lags of highest peak in output array */
+    lags[0] = lags1[0];
+    lags[1] = lags1[0];
+    lags[2] = lags2[0];
+    lags[3] = lags2[0];
+  }
+}
+
+RTC_POP_IGNORING_WFRAME_LARGER_THAN()
+
+/* create weighting matrix by orthogonalizing a basis of polynomials of increasing order
+ * t = (0:4)';
+ * A = [t.^0, t.^1, t.^2, t.^3, t.^4];
+ * [Q, dummy] = qr(A);
+ * P.Weight = Q * diag([0, .1, .5, 1, 1]) * Q'; */
+static const double kWeight[5][5] = {
+  { 0.29714285714286,  -0.30857142857143,  -0.05714285714286,   0.05142857142857,  0.01714285714286},
+  {-0.30857142857143,   0.67428571428571,  -0.27142857142857,  -0.14571428571429,  0.05142857142857},
+  {-0.05714285714286,  -0.27142857142857,   0.65714285714286,  -0.27142857142857, -0.05714285714286},
+  { 0.05142857142857,  -0.14571428571429,  -0.27142857142857,   0.67428571428571, -0.30857142857143},
+  { 0.01714285714286,   0.05142857142857,  -0.05714285714286,  -0.30857142857143,  0.29714285714286}
+};
+
+/* second order high-pass filter */
+static void WebRtcIsac_Highpass(const double* in,
+                         double* out,
+                         double* state,
+                         size_t N) {
+  /* create high-pass filter ocefficients
+   * z = 0.998 * exp(j*2*pi*35/8000);
+   * p = 0.94 * exp(j*2*pi*140/8000);
+   * HP_b = [1, -2*real(z), abs(z)^2];
+   * HP_a = [1, -2*real(p), abs(p)^2]; */
+  static const double a_coef[2] = { 1.86864659625574, -0.88360000000000};
+  static const double b_coef[2] = {-1.99524591718270,  0.99600400000000};
+
+  size_t k;
+
+  for (k=0; k<N; k++) {
+    *out = *in + state[1];
+    state[1] = state[0] + b_coef[0] * *in + a_coef[0] * *out;
+    state[0] = b_coef[1] * *in++ + a_coef[1] * *out++;
+  }
+}
+
+RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
+
+void WebRtcIsac_PitchAnalysis(const double *in,               /* PITCH_FRAME_LEN samples */
+                              double *out,                    /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+                              PitchAnalysisStruct *State,
+                              double *lags,
+                              double *gains)
+{
+  double HPin[PITCH_FRAME_LEN];
+  double Weighted[PITCH_FRAME_LEN];
+  double Whitened[PITCH_FRAME_LEN + QLOOKAHEAD];
+  double inbuf[PITCH_FRAME_LEN + QLOOKAHEAD];
+  double out_G[PITCH_FRAME_LEN + QLOOKAHEAD];          // could be removed by using out instead
+  double out_dG[4][PITCH_FRAME_LEN + QLOOKAHEAD];
+  double old_lag, old_gain;
+  double nrg_wht, tmp;
+  double Wnrg, Wfluct, Wgain;
+  double H[4][4];
+  double grad[4];
+  double dG[4];
+  int k, m, n, iter;
+
+  /* high pass filtering using second order pole-zero filter */
+  WebRtcIsac_Highpass(in, HPin, State->hp_state, PITCH_FRAME_LEN);
+
+  /* copy from state into buffer */
+  memcpy(Whitened, State->whitened_buf, sizeof(double) * QLOOKAHEAD);
+
+  /* compute weighted and whitened signals */
+  WebRtcIsac_WeightingFilter(HPin, &Weighted[0], &Whitened[QLOOKAHEAD], &(State->Wghtstr));
+
+  /* copy from buffer into state */
+  memcpy(State->whitened_buf, Whitened+PITCH_FRAME_LEN, sizeof(double) * QLOOKAHEAD);
+
+  old_lag = State->PFstr_wght.oldlagp[0];
+  old_gain = State->PFstr_wght.oldgainp[0];
+
+  /* inital pitch estimate */
+  WebRtcIsac_InitializePitch(Weighted, old_lag, old_gain, State, lags);
+
+
+  /* Iterative optimization of lags - to be done */
+
+  /* compute energy of whitened signal */
+  nrg_wht = 0.0;
+  for (k = 0; k < PITCH_FRAME_LEN + QLOOKAHEAD; k++)
+    nrg_wht += Whitened[k] * Whitened[k];
+
+
+  /* Iterative optimization of gains */
+
+  /* set weights for energy, gain fluctiation, and spectral gain penalty functions */
+  Wnrg = 1.0 / nrg_wht;
+  Wgain = 0.005;
+  Wfluct = 3.0;
+
+  /* set initial gains */
+  for (k = 0; k < 4; k++)
+    gains[k] = PITCH_MAX_GAIN_06;
+
+  /* two iterations should be enough */
+  for (iter = 0; iter < 2; iter++) {
+    /* compute Jacobian of pre-filter output towards gains */
+    WebRtcIsac_PitchfilterPre_gains(Whitened, out_G, out_dG, &(State->PFstr_wght), lags, gains);
+
+    /* gradient and approximate Hessian (lower triangle) for minimizing the filter's output power */
+    for (k = 0; k < 4; k++) {
+      tmp = 0.0;
+      for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
+        tmp += out_G[n] * out_dG[k][n];
+      grad[k] = tmp * Wnrg;
+    }
+    for (k = 0; k < 4; k++) {
+      for (m = 0; m <= k; m++) {
+        tmp = 0.0;
+        for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
+          tmp += out_dG[m][n] * out_dG[k][n];
+        H[k][m] = tmp * Wnrg;
+      }
+    }
+
+    /* add gradient and Hessian (lower triangle) for dampening fast gain changes */
+    for (k = 0; k < 4; k++) {
+      tmp = kWeight[k+1][0] * old_gain;
+      for (m = 0; m < 4; m++)
+        tmp += kWeight[k+1][m+1] * gains[m];
+      grad[k] += tmp * Wfluct;
+    }
+    for (k = 0; k < 4; k++) {
+      for (m = 0; m <= k; m++) {
+        H[k][m] += kWeight[k+1][m+1] * Wfluct;
+      }
+    }
+
+    /* add gradient and Hessian for dampening gain */
+    for (k = 0; k < 3; k++) {
+      tmp = 1.0 / (1 - gains[k]);
+      grad[k] += tmp * tmp * Wgain;
+      H[k][k] += 2.0 * tmp * (tmp * tmp * Wgain);
+    }
+    tmp = 1.0 / (1 - gains[3]);
+    grad[3] += 1.33 * (tmp * tmp * Wgain);
+    H[3][3] += 2.66 * tmp * (tmp * tmp * Wgain);
+
+
+    /* compute Cholesky factorization of Hessian
+     * by overwritting the upper triangle; scale factors on diagonal
+     * (for non pc-platforms store the inverse of the diagonals seperately to minimize divisions) */
+    H[0][1] = H[1][0] / H[0][0];
+    H[0][2] = H[2][0] / H[0][0];
+    H[0][3] = H[3][0] / H[0][0];
+    H[1][1] -= H[0][0] * H[0][1] * H[0][1];
+    H[1][2] = (H[2][1] - H[0][1] * H[2][0]) / H[1][1];
+    H[1][3] = (H[3][1] - H[0][1] * H[3][0]) / H[1][1];
+    H[2][2] -= H[0][0] * H[0][2] * H[0][2] + H[1][1] * H[1][2] * H[1][2];
+    H[2][3] = (H[3][2] - H[0][2] * H[3][0] - H[1][2] * H[1][1] * H[1][3]) / H[2][2];
+    H[3][3] -= H[0][0] * H[0][3] * H[0][3] + H[1][1] * H[1][3] * H[1][3] + H[2][2] * H[2][3] * H[2][3];
+
+    /* Compute update as  delta_gains = -inv(H) * grad */
+    /* copy and negate */
+    for (k = 0; k < 4; k++)
+      dG[k] = -grad[k];
+    /* back substitution */
+    dG[1] -= dG[0] * H[0][1];
+    dG[2] -= dG[0] * H[0][2] + dG[1] * H[1][2];
+    dG[3] -= dG[0] * H[0][3] + dG[1] * H[1][3] + dG[2] * H[2][3];
+    /* scale */
+    for (k = 0; k < 4; k++)
+      dG[k] /= H[k][k];
+    /* back substitution */
+    dG[2] -= dG[3] * H[2][3];
+    dG[1] -= dG[3] * H[1][3] + dG[2] * H[1][2];
+    dG[0] -= dG[3] * H[0][3] + dG[2] * H[0][2] + dG[1] * H[0][1];
+
+    /* update gains and check range */
+    for (k = 0; k < 4; k++) {
+      gains[k] += dG[k];
+      if (gains[k] > PITCH_MAX_GAIN)
+        gains[k] = PITCH_MAX_GAIN;
+      else if (gains[k] < 0.0)
+        gains[k] = 0.0;
+    }
+  }
+
+  /* update state for next frame */
+  WebRtcIsac_PitchfilterPre(Whitened, out, &(State->PFstr_wght), lags, gains);
+
+  /* concatenate previous input's end and current input */
+  memcpy(inbuf, State->inbuf, sizeof(double) * QLOOKAHEAD);
+  memcpy(inbuf+QLOOKAHEAD, in, sizeof(double) * PITCH_FRAME_LEN);
+
+  /* lookahead pitch filtering for masking analysis */
+  WebRtcIsac_PitchfilterPre_la(inbuf, out, &(State->PFstr), lags, gains);
+
+  /* store last part of input */
+  for (k = 0; k < QLOOKAHEAD; k++)
+    State->inbuf[k] = inbuf[k + PITCH_FRAME_LEN];
+}
+
+RTC_POP_IGNORING_WFRAME_LARGER_THAN()
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
new file mode 100644
index 0000000000..4ab78c20ad
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * pitch_estimator.h
+ *
+ * Pitch functions
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
+
+#include <stddef.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_PitchAnalysis(
+    const double* in, /* PITCH_FRAME_LEN samples */
+    double* out,      /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
+    PitchAnalysisStruct* State,
+    double* lags,
+    double* gains);
+
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_ */
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
new file mode 100644
index 0000000000..bf03dfff2e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
@@ -0,0 +1,388 @@
+/*
+ *  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 <math.h>
+#include <memory.h>
+#include <stdlib.h>
+
+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h"
+#include "rtc_base/compile_assert_c.h"
+
+/*
+ * We are implementing the following filters;
+ *
+ * Pre-filtering:
+ *   y(z) = x(z) + damper(z) * gain * (x(z) + y(z)) * z ^ (-lag);
+ *
+ * Post-filtering:
+ *   y(z) = x(z) - damper(z) * gain * (x(z) + y(z)) * z ^ (-lag);
+ *
+ * Note that `lag` is a floating number so we perform an interpolation to
+ * obtain the correct `lag`.
+ *
+ */
+
+static const double kDampFilter[PITCH_DAMPORDER] = {-0.07, 0.25, 0.64, 0.25,
+    -0.07};
+
+/* interpolation coefficients; generated by design_pitch_filter.m */
+static const double kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
+    {-0.02239172458614,  0.06653315052934, -0.16515880017569,  0.60701333734125,
+     0.64671399919202, -0.20249000396417,  0.09926548334755, -0.04765933793109,
+     0.01754159521746},
+    {-0.01985640750434,  0.05816126837866, -0.13991265473714,  0.44560418147643,
+     0.79117042386876, -0.20266133815188,  0.09585268418555, -0.04533310458084,
+     0.01654127246314},
+    {-0.01463300534216,  0.04229888475060, -0.09897034715253,  0.28284326017787,
+     0.90385267956632, -0.16976950138649,  0.07704272393639, -0.03584218578311,
+     0.01295781500709},
+    {-0.00764851320885,  0.02184035544377, -0.04985561057281,  0.13083306574393,
+     0.97545011664662, -0.10177807997561,  0.04400901776474, -0.02010737175166,
+     0.00719783432422},
+    {-0.00000000000000,  0.00000000000000, -0.00000000000001,  0.00000000000001,
+     0.99999999999999,  0.00000000000001, -0.00000000000001,  0.00000000000000,
+     -0.00000000000000},
+    {0.00719783432422, -0.02010737175166,  0.04400901776474, -0.10177807997562,
+     0.97545011664663,  0.13083306574393, -0.04985561057280,  0.02184035544377,
+     -0.00764851320885},
+    {0.01295781500710, -0.03584218578312,  0.07704272393640, -0.16976950138650,
+     0.90385267956634,  0.28284326017785, -0.09897034715252,  0.04229888475059,
+     -0.01463300534216},
+    {0.01654127246315, -0.04533310458085,  0.09585268418557, -0.20266133815190,
+     0.79117042386878,  0.44560418147640, -0.13991265473712,  0.05816126837865,
+     -0.01985640750433}
+};
+
+/*
+ * Enumerating the operation of the filter.
+ * iSAC has 4 different pitch-filter which are very similar in their structure.
+ *
+ * kPitchFilterPre     : In this mode the filter is operating as pitch
+ *                       pre-filter. This is used at the encoder.
+ * kPitchFilterPost    : In this mode the filter is operating as pitch
+ *                       post-filter. This is the inverse of pre-filter and used
+ *                       in the decoder.
+ * kPitchFilterPreLa   : This is, in structure, similar to pre-filtering but
+ *                       utilizing 3 millisecond lookahead. It is used to
+ *                       obtain the signal for LPC analysis.
+ * kPitchFilterPreGain : This is, in structure, similar to pre-filtering but
+ *                       differential changes in gain is considered. This is
+ *                       used to find the optimal gain.
+ */
+typedef enum {
+  kPitchFilterPre, kPitchFilterPost, kPitchFilterPreLa, kPitchFilterPreGain
+} PitchFilterOperation;
+
+/*
+ * Structure with parameters used for pitch-filtering.
+ * buffer           : a buffer where the sum of previous inputs and outputs
+ *                    are stored.
+ * damper_state     : the state of the damping filter. The filter is defined by
+ *                    `kDampFilter`.
+ * interpol_coeff   : pointer to a set of coefficient which are used to utilize
+ *                    fractional pitch by interpolation.
+ * gain             : pitch-gain to be applied to the current segment of input.
+ * lag              : pitch-lag for the current segment of input.
+ * lag_offset       : the offset of lag w.r.t. current sample.
+ * sub_frame        : sub-frame index, there are 4 pitch sub-frames in an iSAC
+ *                    frame.
+ *                    This specifies the usage of the filter. See
+ *                    'PitchFilterOperation' for operational modes.
+ * num_samples      : number of samples to be processed in each segment.
+ * index            : index of the input and output sample.
+ * damper_state_dg  : state of damping filter for different trial gains.
+ * gain_mult        : differential changes to gain.
+ */
+typedef struct {
+  double buffer[PITCH_INTBUFFSIZE + QLOOKAHEAD];
+  double damper_state[PITCH_DAMPORDER];
+  const double *interpol_coeff;
+  double gain;
+  double lag;
+  int lag_offset;
+
+  int sub_frame;
+  PitchFilterOperation mode;
+  int num_samples;
+  int index;
+
+  double damper_state_dg[4][PITCH_DAMPORDER];
+  double gain_mult[4];
+} PitchFilterParam;
+
+/**********************************************************************
+ * FilterSegment()
+ * Filter one segment, a quarter of a frame.
+ *
+ * Inputs
+ *   in_data      : pointer to the input signal of 30 ms at 8 kHz sample-rate.
+ *   filter_param : pitch filter parameters.
+ *
+ * Outputs
+ *   out_data     : pointer to a buffer where the filtered signal is written to.
+ *   out_dg       : [only used in kPitchFilterPreGain] pointer to a buffer
+ *                  where the output of different gain values (differential
+ *                  change to gain) is written.
+ */
+static void FilterSegment(const double* in_data, PitchFilterParam* parameters,
+                          double* out_data,
+                          double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD]) {
+  int n;
+  int m;
+  int j;
+  double sum;
+  double sum2;
+  /* Index of `parameters->buffer` where the output is written to. */
+  int pos = parameters->index + PITCH_BUFFSIZE;
+  /* Index of `parameters->buffer` where samples are read for fractional-lag
+   * computation. */
+  int pos_lag = pos - parameters->lag_offset;
+
+  for (n = 0; n < parameters->num_samples; ++n) {
+    /* Shift low pass filter states. */
+    for (m = PITCH_DAMPORDER - 1; m > 0; --m) {
+      parameters->damper_state[m] = parameters->damper_state[m - 1];
+    }
+    /* Filter to get fractional pitch. */
+    sum = 0.0;
+    for (m = 0; m < PITCH_FRACORDER; ++m) {
+      sum += parameters->buffer[pos_lag + m] * parameters->interpol_coeff[m];
+    }
+    /* Multiply with gain. */
+    parameters->damper_state[0] = parameters->gain * sum;
+
+    if (parameters->mode == kPitchFilterPreGain) {
+      int lag_index = parameters->index - parameters->lag_offset;
+      int m_tmp = (lag_index < 0) ? -lag_index : 0;
+      /* Update the damper state for the new sample. */
+      for (m = PITCH_DAMPORDER - 1; m > 0; --m) {
+        for (j = 0; j < 4; ++j) {
+          parameters->damper_state_dg[j][m] =
+              parameters->damper_state_dg[j][m - 1];
+        }
+      }
+
+      for (j = 0; j < parameters->sub_frame + 1; ++j) {
+        /* Filter for fractional pitch. */
+        sum2 = 0.0;
+        for (m = PITCH_FRACORDER-1; m >= m_tmp; --m) {
+          /* `lag_index + m` is always larger than or equal to zero, see how
+           * m_tmp is computed. This is equivalent to assume samples outside
+           * `out_dg[j]` are zero. */
+          sum2 += out_dg[j][lag_index + m] * parameters->interpol_coeff[m];
+        }
+        /* Add the contribution of differential gain change. */
+        parameters->damper_state_dg[j][0] = parameters->gain_mult[j] * sum +
+            parameters->gain * sum2;
+      }
+
+      /* Filter with damping filter, and store the results. */
+      for (j = 0; j < parameters->sub_frame + 1; ++j) {
+        sum = 0.0;
+        for (m = 0; m < PITCH_DAMPORDER; ++m) {
+          sum -= parameters->damper_state_dg[j][m] * kDampFilter[m];
+        }
+        out_dg[j][parameters->index] = sum;
+      }
+    }
+    /* Filter with damping filter. */
+    sum = 0.0;
+    for (m = 0; m < PITCH_DAMPORDER; ++m) {
+      sum += parameters->damper_state[m] * kDampFilter[m];
+    }
+
+    /* Subtract from input and update buffer. */
+    out_data[parameters->index] = in_data[parameters->index] - sum;
+    parameters->buffer[pos] = in_data[parameters->index] +
+        out_data[parameters->index];
+
+    ++parameters->index;
+    ++pos;
+    ++pos_lag;
+  }
+  return;
+}
+
+/* Update filter parameters based on the pitch-gains and pitch-lags. */
+static void Update(PitchFilterParam* parameters) {
+  double fraction;
+  int fraction_index;
+  /* Compute integer lag-offset. */
+  parameters->lag_offset = WebRtcIsac_lrint(parameters->lag + PITCH_FILTDELAY +
+                                            0.5);
+  /* Find correct set of coefficients for computing fractional pitch. */
+  fraction = parameters->lag_offset - (parameters->lag + PITCH_FILTDELAY);
+  fraction_index = WebRtcIsac_lrint(PITCH_FRACS * fraction - 0.5);
+  parameters->interpol_coeff = kIntrpCoef[fraction_index];
+
+  if (parameters->mode == kPitchFilterPreGain) {
+    /* If in this mode make a differential change to pitch gain. */
+    parameters->gain_mult[parameters->sub_frame] += 0.2;
+    if (parameters->gain_mult[parameters->sub_frame] > 1.0) {
+      parameters->gain_mult[parameters->sub_frame] = 1.0;
+    }
+    if (parameters->sub_frame > 0) {
+      parameters->gain_mult[parameters->sub_frame - 1] -= 0.2;
+    }
+  }
+}
+
+/******************************************************************************
+ * FilterFrame()
+ * Filter a frame of 30 millisecond, given pitch-lags and pitch-gains.
+ *
+ * Inputs
+ *   in_data     : pointer to the input signal of 30 ms at 8 kHz sample-rate.
+ *   lags        : pointer to pitch-lags, 4 lags per frame.
+ *   gains       : pointer to pitch-gians, 4 gains per frame.
+ *   mode        : defining the functionality of the filter. It takes the
+ *                 following values.
+ *                 kPitchFilterPre:     Pitch pre-filter, used at encoder.
+ *                 kPitchFilterPost:    Pitch post-filter, used at decoder.
+ *                 kPitchFilterPreLa:   Pitch pre-filter with lookahead.
+ *                 kPitchFilterPreGain: Pitch pre-filter used to otain optimal
+ *                                      pitch-gains.
+ *
+ * Outputs
+ *   out_data    : pointer to a buffer where the filtered signal is written to.
+ *   out_dg      : [only used in kPitchFilterPreGain] pointer to a buffer
+ *                 where the output of different gain values (differential
+ *                 change to gain) is written.
+ */
+static void FilterFrame(const double* in_data, PitchFiltstr* filter_state,
+                        double* lags, double* gains, PitchFilterOperation mode,
+                        double* out_data,
+                        double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD]) {
+  PitchFilterParam filter_parameters;
+  double gain_delta, lag_delta;
+  double old_lag, old_gain;
+  int n;
+  int m;
+  const double kEnhancer = 1.3;
+
+  /* Set up buffer and states. */
+  filter_parameters.index = 0;
+  filter_parameters.lag_offset = 0;
+  filter_parameters.mode = mode;
+  /* Copy states to local variables. */
+  memcpy(filter_parameters.buffer, filter_state->ubuf,
+         sizeof(filter_state->ubuf));
+  RTC_COMPILE_ASSERT(sizeof(filter_parameters.buffer) >=
+                 sizeof(filter_state->ubuf));
+  memset(filter_parameters.buffer +
+             sizeof(filter_state->ubuf) / sizeof(filter_state->ubuf[0]),
+         0, sizeof(filter_parameters.buffer) - sizeof(filter_state->ubuf));
+  memcpy(filter_parameters.damper_state, filter_state->ystate,
+         sizeof(filter_state->ystate));
+
+  if (mode == kPitchFilterPreGain) {
+    /* Clear buffers. */
+    memset(filter_parameters.gain_mult, 0, sizeof(filter_parameters.gain_mult));
+    memset(filter_parameters.damper_state_dg, 0,
+           sizeof(filter_parameters.damper_state_dg));
+    for (n = 0; n < PITCH_SUBFRAMES; ++n) {
+      //memset(out_dg[n], 0, sizeof(double) * (PITCH_FRAME_LEN + QLOOKAHEAD));
+      memset(out_dg[n], 0, sizeof(out_dg[n]));
+    }
+  } else if (mode == kPitchFilterPost) {
+    /* Make output more periodic. Negative sign is to change the structure
+     * of the filter. */
+    for (n = 0; n < PITCH_SUBFRAMES; ++n) {
+      gains[n] *= -kEnhancer;
+    }
+  }
+
+  old_lag = *filter_state->oldlagp;
+  old_gain = *filter_state->oldgainp;
+
+  /* No interpolation if pitch lag step is big. */
+  if ((lags[0] > (PITCH_UPSTEP * old_lag)) ||
+      (lags[0] < (PITCH_DOWNSTEP * old_lag))) {
+    old_lag = lags[0];
+    old_gain = gains[0];
+
+    if (mode == kPitchFilterPreGain) {
+      filter_parameters.gain_mult[0] = 1.0;
+    }
+  }
+
+  filter_parameters.num_samples = PITCH_UPDATE;
+  for (m = 0; m < PITCH_SUBFRAMES; ++m) {
+    /* Set the sub-frame value. */
+    filter_parameters.sub_frame = m;
+    /* Calculate interpolation steps for pitch-lag and pitch-gain. */
+    lag_delta = (lags[m] - old_lag) / PITCH_GRAN_PER_SUBFRAME;
+    filter_parameters.lag = old_lag;
+    gain_delta = (gains[m] - old_gain) / PITCH_GRAN_PER_SUBFRAME;
+    filter_parameters.gain = old_gain;
+    /* Store for the next sub-frame. */
+    old_lag = lags[m];
+    old_gain = gains[m];
+
+    for (n = 0; n < PITCH_GRAN_PER_SUBFRAME; ++n) {
+      /* Step-wise interpolation of pitch gains and lags. As pitch-lag changes,
+       * some parameters of filter need to be update. */
+      filter_parameters.gain += gain_delta;
+      filter_parameters.lag += lag_delta;
+      /* Update parameters according to new lag value. */
+      Update(&filter_parameters);
+      /* Filter a segment of input. */
+      FilterSegment(in_data, &filter_parameters, out_data, out_dg);
+    }
+  }
+
+  if (mode != kPitchFilterPreGain) {
+    /* Export buffer and states. */
+    memcpy(filter_state->ubuf, &filter_parameters.buffer[PITCH_FRAME_LEN],
+           sizeof(filter_state->ubuf));
+    memcpy(filter_state->ystate, filter_parameters.damper_state,
+           sizeof(filter_state->ystate));
+
+    /* Store for the next frame. */
+    *filter_state->oldlagp = old_lag;
+    *filter_state->oldgainp = old_gain;
+  }
+
+  if ((mode == kPitchFilterPreGain) || (mode == kPitchFilterPreLa)) {
+    /* Filter the lookahead segment, this is treated as the last sub-frame. So
+     * set `pf_param` to last sub-frame. */
+    filter_parameters.sub_frame = PITCH_SUBFRAMES - 1;
+    filter_parameters.num_samples = QLOOKAHEAD;
+    FilterSegment(in_data, &filter_parameters, out_data, out_dg);
+  }
+}
+
+void WebRtcIsac_PitchfilterPre(double* in_data, double* out_data,
+                               PitchFiltstr* pf_state, double* lags,
+                               double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPre, out_data, NULL);
+}
+
+void WebRtcIsac_PitchfilterPre_la(double* in_data, double* out_data,
+                                  PitchFiltstr* pf_state, double* lags,
+                                  double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPreLa, out_data,
+              NULL);
+}
+
+void WebRtcIsac_PitchfilterPre_gains(
+    double* in_data, double* out_data,
+    double out_dg[][PITCH_FRAME_LEN + QLOOKAHEAD], PitchFiltstr *pf_state,
+    double* lags, double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPreGain, out_data,
+              out_dg);
+}
+
+void WebRtcIsac_PitchfilterPost(double* in_data, double* out_data,
+                                PitchFiltstr* pf_state, double* lags,
+                                double* gains) {
+  FilterFrame(in_data, pf_state, lags, gains, kPitchFilterPost, out_data, NULL);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.h
new file mode 100644
index 0000000000..9a232de87b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
+
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
+
+void WebRtcIsac_PitchfilterPre(double* indat,
+                               double* outdat,
+                               PitchFiltstr* pfp,
+                               double* lags,
+                               double* gains);
+
+void WebRtcIsac_PitchfilterPost(double* indat,
+                                double* outdat,
+                                PitchFiltstr* pfp,
+                                double* lags,
+                                double* gains);
+
+void WebRtcIsac_PitchfilterPre_la(double* indat,
+                                  double* outdat,
+                                  PitchFiltstr* pfp,
+                                  double* lags,
+                                  double* gains);
+
+void WebRtcIsac_PitchfilterPre_gains(
+    double* indat,
+    double* outdat,
+    double out_dG[][PITCH_FRAME_LEN + QLOOKAHEAD],
+    PitchFiltstr* pfp,
+    double* lags,
+    double* gains);
+
+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/settings.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/settings.h
new file mode 100644
index 0000000000..abce90c4f5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/settings.h
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * settings.h
+ *
+ * Declaration of #defines used in the iSAC codec
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
+
+/* sampling frequency (Hz) */
+#define FS 16000
+
+/* number of samples per frame (either 320 (20ms), 480 (30ms) or 960 (60ms)) */
+#define INITIAL_FRAMESAMPLES 960
+
+/* do not modify the following; this will have to be modified if we
+ * have a 20ms framesize option */
+/**********************************************************************/
+/* miliseconds */
+#define FRAMESIZE 30
+/* number of samples per frame processed in the encoder, 480 */
+#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
+#define FRAMESAMPLES_HALF 240
+#define FRAMESAMPLES_QUARTER 120
+/**********************************************************************/
+
+/* max number of samples per frame (= 60 ms frame) */
+#define MAX_FRAMESAMPLES 960
+#define MAX_SWBFRAMESAMPLES (MAX_FRAMESAMPLES * 2)
+/* number of samples per 10ms frame */
+#define FRAMESAMPLES_10ms ((10 * FS) / 1000)
+#define SWBFRAMESAMPLES_10ms (FRAMESAMPLES_10ms * 2)
+/* number of samples in 30 ms frame */
+#define FRAMESAMPLES_30ms 480
+/* number of subframes */
+#define SUBFRAMES 6
+/* length of a subframe */
+#define UPDATE 80
+/* length of half a subframe (low/high band) */
+#define HALF_SUBFRAMELEN (UPDATE / 2)
+/* samples of look ahead (in a half-band, so actually
+ * half the samples of look ahead @ FS) */
+#define QLOOKAHEAD 24 /* 3 ms */
+/* order of AR model in spectral entropy coder */
+#define AR_ORDER 6
+/* order of LP model in spectral entropy coder */
+#define LP_ORDER 0
+
+/* window length (masking analysis) */
+#define WINLEN 256
+/* order of low-band pole filter used to approximate masking curve */
+#define ORDERLO 12
+/* order of hi-band pole filter used to approximate masking curve */
+#define ORDERHI 6
+
+#define UB_LPC_ORDER 4
+#define UB_LPC_VEC_PER_FRAME 2
+#define UB16_LPC_VEC_PER_FRAME 4
+#define UB_ACTIVE_SUBFRAMES 2
+#define UB_MAX_LPC_ORDER 6
+#define UB_INTERPOL_SEGMENTS 1
+#define UB16_INTERPOL_SEGMENTS 3
+#define LB_TOTAL_DELAY_SAMPLES 48
+enum ISACBandwidth { isac8kHz = 8, isac12kHz = 12, isac16kHz = 16 };
+enum ISACBand {
+  kIsacLowerBand = 0,
+  kIsacUpperBand12 = 1,
+  kIsacUpperBand16 = 2
+};
+enum IsacSamplingRate { kIsacWideband = 16, kIsacSuperWideband = 32 };
+#define UB_LPC_GAIN_DIM SUBFRAMES
+#define FB_STATE_SIZE_WORD32 6
+
+/* order for post_filter_bank */
+#define POSTQORDER 3
+/* order for pre-filterbank */
+#define QORDER 3
+/* another order */
+#define QORDER_ALL (POSTQORDER + QORDER - 1)
+/* for decimator */
+#define ALLPASSSECTIONS 2
+
+/* array size for byte stream in number of bytes. */
+/* The old maximum size still needed for the decoding */
+#define STREAM_SIZE_MAX 600
+#define STREAM_SIZE_MAX_30 200 /* 200 bytes=53.4 kbps @ 30 ms.framelength */
+#define STREAM_SIZE_MAX_60 400 /* 400 bytes=53.4 kbps @ 60 ms.framelength */
+
+/* storage size for bit counts */
+#define BIT_COUNTER_SIZE 30
+/* maximum order of any AR model or filter */
+#define MAX_AR_MODEL_ORDER 12  // 50
+
+/* For pitch analysis */
+#define PITCH_FRAME_LEN (FRAMESAMPLES_HALF) /* 30 ms  */
+#define PITCH_MAX_LAG 140                   /* 57 Hz  */
+#define PITCH_MIN_LAG 20                    /* 400 Hz */
+#define PITCH_MAX_GAIN 0.45
+#define PITCH_MAX_GAIN_06 0.27 /* PITCH_MAX_GAIN*0.6 */
+#define PITCH_MAX_GAIN_Q12 1843
+#define PITCH_LAG_SPAN2 (PITCH_MAX_LAG / 2 - PITCH_MIN_LAG / 2 + 5)
+#define PITCH_CORR_LEN2 60 /* 15 ms  */
+#define PITCH_CORR_STEP2 (PITCH_FRAME_LEN / 4)
+#define PITCH_BW 11 /* half the band width of correlation surface */
+#define PITCH_SUBFRAMES 4
+#define PITCH_GRAN_PER_SUBFRAME 5
+#define PITCH_SUBFRAME_LEN (PITCH_FRAME_LEN / PITCH_SUBFRAMES)
+#define PITCH_UPDATE (PITCH_SUBFRAME_LEN / PITCH_GRAN_PER_SUBFRAME)
+/* maximum number of peaks to be examined in correlation surface */
+#define PITCH_MAX_NUM_PEAKS 10
+#define PITCH_PEAK_DECAY 0.85
+/* For weighting filter */
+#define PITCH_WLPCORDER 6
+#define PITCH_WLPCWINLEN PITCH_FRAME_LEN
+#define PITCH_WLPCASYM 0.3 /* asymmetry parameter */
+#define PITCH_WLPCBUFLEN PITCH_WLPCWINLEN
+/* For pitch filter */
+/* Extra 50 for fraction and LP filters */
+#define PITCH_BUFFSIZE (PITCH_MAX_LAG + 50)
+#define PITCH_INTBUFFSIZE (PITCH_FRAME_LEN + PITCH_BUFFSIZE)
+/* Max rel. step for interpolation */
+#define PITCH_UPSTEP 1.5
+/* Max rel. step for interpolation */
+#define PITCH_DOWNSTEP 0.67
+#define PITCH_FRACS 8
+#define PITCH_FRACORDER 9
+#define PITCH_DAMPORDER 5
+#define PITCH_FILTDELAY 1.5f
+/* stepsize for quantization of the pitch Gain */
+#define PITCH_GAIN_STEPSIZE 0.125
+
+/* Order of high pass filter */
+#define HPORDER 2
+
+/* some mathematical constants */
+/* log2(exp) */
+#define LOG2EXP 1.44269504088896
+#define PI 3.14159265358979
+
+/* Maximum number of iterations allowed to limit payload size */
+#define MAX_PAYLOAD_LIMIT_ITERATION 5
+
+/* Redundant Coding */
+#define RCU_BOTTLENECK_BPS 16000
+#define RCU_TRANSCODING_SCALE 0.40f
+#define RCU_TRANSCODING_SCALE_INVERSE 2.5f
+
+#define RCU_TRANSCODING_SCALE_UB 0.50f
+#define RCU_TRANSCODING_SCALE_UB_INVERSE 2.0f
+
+/* Define Error codes */
+/* 6000 General */
+#define ISAC_MEMORY_ALLOCATION_FAILED 6010
+#define ISAC_MODE_MISMATCH 6020
+#define ISAC_DISALLOWED_BOTTLENECK 6030
+#define ISAC_DISALLOWED_FRAME_LENGTH 6040
+#define ISAC_UNSUPPORTED_SAMPLING_FREQUENCY 6050
+
+/* 6200 Bandwidth estimator */
+#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
+/* 6400 Encoder */
+#define ISAC_ENCODER_NOT_INITIATED 6410
+#define ISAC_DISALLOWED_CODING_MODE 6420
+#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
+#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
+#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
+#define ISAC_DISALLOWED_ENCODER_BANDWIDTH 6460
+/* 6600 Decoder */
+#define ISAC_DECODER_NOT_INITIATED 6610
+#define ISAC_EMPTY_PACKET 6620
+#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
+#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
+#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
+#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
+#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
+#define ISAC_RANGE_ERROR_DECODE_LPC 6680
+#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
+#define ISAC_LENGTH_MISMATCH 6730
+#define ISAC_RANGE_ERROR_DECODE_BANDWITH 6740
+#define ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER 6750
+#define ISAC_DISALLOWED_LPC_MODEL 6760
+/* 6800 Call setup formats */
+#define ISAC_INCOMPATIBLE_FORMATS 6810
+
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_ */
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/structs.h b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/structs.h
new file mode 100644
index 0000000000..6861ca42bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/isac/main/source/structs.h
@@ -0,0 +1,448 @@
+/*
+ *  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.
+ */
+
+/*
+ * structs.h
+ *
+ * This header file contains all the structs used in the ISAC codec
+ *
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+
+#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
+#include "modules/third_party/fft/fft.h"
+
+typedef struct Bitstreamstruct {
+  uint8_t stream[STREAM_SIZE_MAX];
+  uint32_t W_upper;
+  uint32_t streamval;
+  uint32_t stream_index;
+
+} Bitstr;
+
+typedef struct {
+  double DataBufferLo[WINLEN];
+  double DataBufferHi[WINLEN];
+
+  double CorrBufLo[ORDERLO + 1];
+  double CorrBufHi[ORDERHI + 1];
+
+  float PreStateLoF[ORDERLO + 1];
+  float PreStateLoG[ORDERLO + 1];
+  float PreStateHiF[ORDERHI + 1];
+  float PreStateHiG[ORDERHI + 1];
+  float PostStateLoF[ORDERLO + 1];
+  float PostStateLoG[ORDERLO + 1];
+  float PostStateHiF[ORDERHI + 1];
+  float PostStateHiG[ORDERHI + 1];
+
+  double OldEnergy;
+
+} MaskFiltstr;
+
+typedef struct {
+  // state vectors for each of the two analysis filters
+  double INSTAT1[2 * (QORDER - 1)];
+  double INSTAT2[2 * (QORDER - 1)];
+  double INSTATLA1[2 * (QORDER - 1)];
+  double INSTATLA2[2 * (QORDER - 1)];
+  double INLABUF1[QLOOKAHEAD];
+  double INLABUF2[QLOOKAHEAD];
+
+  float INSTAT1_float[2 * (QORDER - 1)];
+  float INSTAT2_float[2 * (QORDER - 1)];
+  float INSTATLA1_float[2 * (QORDER - 1)];
+  float INSTATLA2_float[2 * (QORDER - 1)];
+  float INLABUF1_float[QLOOKAHEAD];
+  float INLABUF2_float[QLOOKAHEAD];
+
+  /* High pass filter */
+  double HPstates[HPORDER];
+  float HPstates_float[HPORDER];
+
+} PreFiltBankstr;
+
+typedef struct {
+  // state vectors for each of the two analysis filters
+  double STATE_0_LOWER[2 * POSTQORDER];
+  double STATE_0_UPPER[2 * POSTQORDER];
+
+  /* High pass filter */
+  double HPstates1[HPORDER];
+  double HPstates2[HPORDER];
+
+  float STATE_0_LOWER_float[2 * POSTQORDER];
+  float STATE_0_UPPER_float[2 * POSTQORDER];
+
+  float HPstates1_float[HPORDER];
+  float HPstates2_float[HPORDER];
+
+} PostFiltBankstr;
+
+typedef struct {
+  // data buffer for pitch filter
+  double ubuf[PITCH_BUFFSIZE];
+
+  // low pass state vector
+  double ystate[PITCH_DAMPORDER];
+
+  // old lag and gain
+  double oldlagp[1];
+  double oldgainp[1];
+
+} PitchFiltstr;
+
+typedef struct {
+  // data buffer
+  double buffer[PITCH_WLPCBUFLEN];
+
+  // state vectors
+  double istate[PITCH_WLPCORDER];
+  double weostate[PITCH_WLPCORDER];
+  double whostate[PITCH_WLPCORDER];
+
+  // LPC window   -> should be a global array because constant
+  double window[PITCH_WLPCWINLEN];
+
+} WeightFiltstr;
+
+typedef struct {
+  // for inital estimator
+  double dec_buffer[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
+                    PITCH_FRAME_LEN / 2 + 2];
+  double decimator_state[2 * ALLPASSSECTIONS + 1];
+  double hp_state[2];
+
+  double whitened_buf[QLOOKAHEAD];
+
+  double inbuf[QLOOKAHEAD];
+
+  PitchFiltstr PFstr_wght;
+  PitchFiltstr PFstr;
+  WeightFiltstr Wghtstr;
+
+} PitchAnalysisStruct;
+
+/* Have instance of struct together with other iSAC structs */
+typedef struct {
+  /* Previous frame length (in ms)                                    */
+  int32_t prev_frame_length;
+
+  /* Previous RTP timestamp from received
+     packet (in samples relative beginning)                           */
+  int32_t prev_rec_rtp_number;
+
+  /* Send timestamp for previous packet (in ms using timeGetTime())   */
+  uint32_t prev_rec_send_ts;
+
+  /* Arrival time for previous packet (in ms using timeGetTime())     */
+  uint32_t prev_rec_arr_ts;
+
+  /* rate of previous packet, derived from RTP timestamps (in bits/s) */
+  float prev_rec_rtp_rate;
+
+  /* Time sinse the last update of the BN estimate (in ms)            */
+  uint32_t last_update_ts;
+
+  /* Time sinse the last reduction (in ms)                            */
+  uint32_t last_reduction_ts;
+
+  /* How many times the estimate was update in the beginning          */
+  int32_t count_tot_updates_rec;
+
+  /* The estimated bottle neck rate from there to here (in bits/s)    */
+  int32_t rec_bw;
+  float rec_bw_inv;
+  float rec_bw_avg;
+  float rec_bw_avg_Q;
+
+  /* The estimated mean absolute jitter value,
+     as seen on this side (in ms)                                     */
+  float rec_jitter;
+  float rec_jitter_short_term;
+  float rec_jitter_short_term_abs;
+  float rec_max_delay;
+  float rec_max_delay_avg_Q;
+
+  /* (assumed) bitrate for headers (bps)                              */
+  float rec_header_rate;
+
+  /* The estimated bottle neck rate from here to there (in bits/s)    */
+  float send_bw_avg;
+
+  /* The estimated mean absolute jitter value, as seen on
+     the other siee (in ms)                                           */
+  float send_max_delay_avg;
+
+  // number of packets received since last update
+  int num_pkts_rec;
+
+  int num_consec_rec_pkts_over_30k;
+
+  // flag for marking that a high speed network has been
+  // detected downstream
+  int hsn_detect_rec;
+
+  int num_consec_snt_pkts_over_30k;
+
+  // flag for marking that a high speed network has
+  // been detected upstream
+  int hsn_detect_snd;
+
+  uint32_t start_wait_period;
+
+  int in_wait_period;
+
+  int change_to_WB;
+
+  uint32_t senderTimestamp;
+  uint32_t receiverTimestamp;
+  // enum IsacSamplingRate incomingStreamSampFreq;
+  uint16_t numConsecLatePkts;
+  float consecLatency;
+  int16_t inWaitLatePkts;
+
+  IsacBandwidthInfo external_bw_info;
+} BwEstimatorstr;
+
+typedef struct {
+  /* boolean, flags if previous packet exceeded B.N. */
+  int PrevExceed;
+  /* ms */
+  int ExceedAgo;
+  /* packets left to send in current burst */
+  int BurstCounter;
+  /* packets */
+  int InitCounter;
+  /* ms remaining in buffer when next packet will be sent */
+  double StillBuffered;
+
+} RateModel;
+
+/* The following strutc is used to store data from encoding, to make it
+   fast and easy to construct a new bitstream with a different Bandwidth
+   estimate. All values (except framelength and minBytes) is double size to
+   handle 60 ms of data.
+*/
+typedef struct {
+  /* Used to keep track of if it is first or second part of 60 msec packet */
+  int startIdx;
+
+  /* Frame length in samples */
+  int16_t framelength;
+
+  /* Pitch Gain */
+  int pitchGain_index[2];
+
+  /* Pitch Lag */
+  double meanGain[2];
+  int pitchIndex[PITCH_SUBFRAMES * 2];
+
+  /* LPC */
+  int LPCindex_s[108 * 2]; /* KLT_ORDER_SHAPE = 108 */
+  int LPCindex_g[12 * 2];  /* KLT_ORDER_GAIN = 12 */
+  double LPCcoeffs_lo[(ORDERLO + 1) * SUBFRAMES * 2];
+  double LPCcoeffs_hi[(ORDERHI + 1) * SUBFRAMES * 2];
+
+  /* Encode Spec */
+  int16_t fre[FRAMESAMPLES];
+  int16_t fim[FRAMESAMPLES];
+  int16_t AvgPitchGain[2];
+
+  /* Used in adaptive mode only */
+  int minBytes;
+
+} IsacSaveEncoderData;
+
+typedef struct {
+  int indexLPCShape[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
+  double lpcGain[SUBFRAMES << 1];
+  int lpcGainIndex[SUBFRAMES << 1];
+
+  Bitstr bitStreamObj;
+
+  int16_t realFFT[FRAMESAMPLES_HALF];
+  int16_t imagFFT[FRAMESAMPLES_HALF];
+} ISACUBSaveEncDataStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PreFiltBankstr prefiltbankstr_obj;
+  PitchFiltstr pitchfiltstr_obj;
+  PitchAnalysisStruct pitchanalysisstr_obj;
+  FFTstr fftstr_obj;
+  IsacSaveEncoderData SaveEnc_obj;
+
+  int buffer_index;
+  int16_t current_framesamples;
+
+  float data_buffer_float[FRAMESAMPLES_30ms];
+
+  int frame_nb;
+  double bottleneck;
+  int16_t new_framelength;
+  double s2nr;
+
+  /* Maximum allowed number of bits for a 30 msec packet */
+  int16_t payloadLimitBytes30;
+  /* Maximum allowed number of bits for a 30 msec packet */
+  int16_t payloadLimitBytes60;
+  /* Maximum allowed number of bits for both 30 and 60 msec packet */
+  int16_t maxPayloadBytes;
+  /* Maximum allowed rate in bytes per 30 msec packet */
+  int16_t maxRateInBytes;
+
+  /*---
+    If set to 1 iSAC will not adapt the frame-size, if used in
+    channel-adaptive mode. The initial value will be used for all rates.
+    ---*/
+  int16_t enforceFrameSize;
+
+  /*-----
+    This records the BWE index the encoder injected into the bit-stream.
+    It will be used in RCU. The same BWE index of main payload will be in
+    the redundant payload. We can not retrieve it from BWE because it is
+    a recursive procedure (WebRtcIsac_GetDownlinkBwJitIndexImpl) and has to be
+    called only once per each encode.
+    -----*/
+  int16_t lastBWIdx;
+} ISACLBEncStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PreFiltBankstr prefiltbankstr_obj;
+  FFTstr fftstr_obj;
+  ISACUBSaveEncDataStruct SaveEnc_obj;
+
+  int buffer_index;
+  float data_buffer_float[MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES];
+  double bottleneck;
+  /* Maximum allowed number of bits for a 30 msec packet */
+  // int16_t        payloadLimitBytes30;
+  /* Maximum allowed number of bits for both 30 and 60 msec packet */
+  // int16_t        maxPayloadBytes;
+  int16_t maxPayloadSizeBytes;
+
+  double lastLPCVec[UB_LPC_ORDER];
+  int16_t numBytesUsed;
+  int16_t lastJitterInfo;
+} ISACUBEncStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PostFiltBankstr postfiltbankstr_obj;
+  PitchFiltstr pitchfiltstr_obj;
+  FFTstr fftstr_obj;
+
+} ISACLBDecStruct;
+
+typedef struct {
+  Bitstr bitstr_obj;
+  MaskFiltstr maskfiltstr_obj;
+  PostFiltBankstr postfiltbankstr_obj;
+  FFTstr fftstr_obj;
+
+} ISACUBDecStruct;
+
+typedef struct {
+  ISACLBEncStruct ISACencLB_obj;
+  ISACLBDecStruct ISACdecLB_obj;
+} ISACLBStruct;
+
+typedef struct {
+  ISACUBEncStruct ISACencUB_obj;
+  ISACUBDecStruct ISACdecUB_obj;
+} ISACUBStruct;
+
+/*
+  This struct is used to take a snapshot of the entropy coder and LPC gains
+  right before encoding LPC gains. This allows us to go back to that state
+  if we like to limit the payload size.
+*/
+typedef struct {
+  /* 6 lower-band & 6 upper-band */
+  double loFiltGain[SUBFRAMES];
+  double hiFiltGain[SUBFRAMES];
+  /* Upper boundary of interval W */
+  uint32_t W_upper;
+  uint32_t streamval;
+  /* Index to the current position in bytestream */
+  uint32_t stream_index;
+  uint8_t stream[3];
+} transcode_obj;
+
+typedef struct {
+  // TODO(kwiberg): The size of these tables could be reduced by storing floats
+  // instead of doubles, and by making use of the identity cos(x) =
+  // sin(x+pi/2). They could also be made global constants that we fill in at
+  // compile time.
+  double costab1[FRAMESAMPLES_HALF];
+  double sintab1[FRAMESAMPLES_HALF];
+  double costab2[FRAMESAMPLES_QUARTER];
+  double sintab2[FRAMESAMPLES_QUARTER];
+} TransformTables;
+
+typedef struct {
+  // lower-band codec instance
+  ISACLBStruct instLB;
+  // upper-band codec instance
+  ISACUBStruct instUB;
+
+  // Bandwidth Estimator and model for the rate.
+  BwEstimatorstr bwestimator_obj;
+  RateModel rate_data_obj;
+  double MaxDelay;
+
+  /* 0 = adaptive; 1 = instantaneous */
+  int16_t codingMode;
+
+  // overall bottleneck of the codec
+  int32_t bottleneck;
+
+  // QMF Filter state
+  int32_t analysisFBState1[FB_STATE_SIZE_WORD32];
+  int32_t analysisFBState2[FB_STATE_SIZE_WORD32];
+  int32_t synthesisFBState1[FB_STATE_SIZE_WORD32];
+  int32_t synthesisFBState2[FB_STATE_SIZE_WORD32];
+
+  // Error Code
+  int16_t errorCode;
+
+  // bandwidth of the encoded audio 8, 12 or 16 kHz
+  enum ISACBandwidth bandwidthKHz;
+  // Sampling rate of audio, encoder and decode,  8 or 16 kHz
+  enum IsacSamplingRate encoderSamplingRateKHz;
+  enum IsacSamplingRate decoderSamplingRateKHz;
+  // Flag to keep track of initializations, lower & upper-band
+  // encoder and decoder.
+  int16_t initFlag;
+
+  // Flag to to indicate signal bandwidth switch
+  int16_t resetFlag_8kHz;
+
+  // Maximum allowed rate, measured in Bytes per 30 ms.
+  int16_t maxRateBytesPer30Ms;
+  // Maximum allowed payload-size, measured in Bytes.
+  int16_t maxPayloadSizeBytes;
+  /* The expected sampling rate of the input signal. Valid values are 16000
+   * and 32000. This is not the operation sampling rate of the codec. */
+  uint16_t in_sample_rate_hz;
+
+  // Trig tables for WebRtcIsac_Time2Spec and WebRtcIsac_Spec2time.
+  TransformTables transform_tables;
+} ISACMainStruct;
+
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
new file mode 100644
index 0000000000..dacf325082
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+LegacyEncodedAudioFrame::LegacyEncodedAudioFrame(AudioDecoder* decoder,
+                                                 rtc::Buffer&& payload)
+    : decoder_(decoder), payload_(std::move(payload)) {}
+
+LegacyEncodedAudioFrame::~LegacyEncodedAudioFrame() = default;
+
+size_t LegacyEncodedAudioFrame::Duration() const {
+  const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+  return (ret < 0) ? 0 : static_cast<size_t>(ret);
+}
+
+absl::optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
+LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
+  AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+  const int ret = decoder_->Decode(
+      payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+      decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+
+  if (ret < 0)
+    return absl::nullopt;
+
+  return DecodeResult{static_cast<size_t>(ret), speech_type};
+}
+
+std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
+    AudioDecoder* decoder,
+    rtc::Buffer&& payload,
+    uint32_t timestamp,
+    size_t bytes_per_ms,
+    uint32_t timestamps_per_ms) {
+  RTC_DCHECK(payload.data());
+  std::vector<AudioDecoder::ParseResult> results;
+  size_t split_size_bytes = payload.size();
+
+  // Find a "chunk size" >= 20 ms and < 40 ms.
+  const size_t min_chunk_size = bytes_per_ms * 20;
+  if (min_chunk_size >= payload.size()) {
+    std::unique_ptr<LegacyEncodedAudioFrame> frame(
+        new LegacyEncodedAudioFrame(decoder, std::move(payload)));
+    results.emplace_back(timestamp, 0, std::move(frame));
+  } else {
+    // Reduce the split size by half as long as `split_size_bytes` is at least
+    // twice the minimum chunk size (so that the resulting size is at least as
+    // large as the minimum chunk size).
+    while (split_size_bytes >= 2 * min_chunk_size) {
+      split_size_bytes /= 2;
+    }
+
+    const uint32_t timestamps_per_chunk = static_cast<uint32_t>(
+        split_size_bytes * timestamps_per_ms / bytes_per_ms);
+    size_t byte_offset;
+    uint32_t timestamp_offset;
+    for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
+         byte_offset += split_size_bytes,
+        timestamp_offset += timestamps_per_chunk) {
+      split_size_bytes =
+          std::min(split_size_bytes, payload.size() - byte_offset);
+      rtc::Buffer new_payload(payload.data() + byte_offset, split_size_bytes);
+      std::unique_ptr<LegacyEncodedAudioFrame> frame(
+          new LegacyEncodedAudioFrame(decoder, std::move(new_payload)));
+      results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
+    }
+  }
+
+  return results;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
new file mode 100644
index 0000000000..21da1367ed
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
+#define MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {
+ public:
+  LegacyEncodedAudioFrame(AudioDecoder* decoder, rtc::Buffer&& payload);
+  ~LegacyEncodedAudioFrame() override;
+
+  static std::vector<AudioDecoder::ParseResult> SplitBySamples(
+      AudioDecoder* decoder,
+      rtc::Buffer&& payload,
+      uint32_t timestamp,
+      size_t bytes_per_ms,
+      uint32_t timestamps_per_ms);
+
+  size_t Duration() const override;
+
+  absl::optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override;
+
+  // For testing:
+  const rtc::Buffer& payload() const { return payload_; }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
new file mode 100644
index 0000000000..f81aeeea80
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+enum class NetEqDecoder {
+  kDecoderPCMu,
+  kDecoderPCMa,
+  kDecoderPCMu_2ch,
+  kDecoderPCMa_2ch,
+  kDecoderPCM16B,
+  kDecoderPCM16Bwb,
+  kDecoderPCM16Bswb32kHz,
+  kDecoderPCM16Bswb48kHz,
+  kDecoderPCM16B_2ch,
+  kDecoderPCM16Bwb_2ch,
+  kDecoderPCM16Bswb32kHz_2ch,
+  kDecoderPCM16Bswb48kHz_2ch,
+  kDecoderPCM16B_5ch,
+  kDecoderG722,
+};
+
+class SplitBySamplesTest : public ::testing::TestWithParam<NetEqDecoder> {
+ protected:
+  virtual void SetUp() {
+    decoder_type_ = GetParam();
+    switch (decoder_type_) {
+      case NetEqDecoder::kDecoderPCMu:
+      case NetEqDecoder::kDecoderPCMa:
+        bytes_per_ms_ = 8;
+        samples_per_ms_ = 8;
+        break;
+      case NetEqDecoder::kDecoderPCMu_2ch:
+      case NetEqDecoder::kDecoderPCMa_2ch:
+        bytes_per_ms_ = 2 * 8;
+        samples_per_ms_ = 8;
+        break;
+      case NetEqDecoder::kDecoderG722:
+        bytes_per_ms_ = 8;
+        samples_per_ms_ = 16;
+        break;
+      case NetEqDecoder::kDecoderPCM16B:
+        bytes_per_ms_ = 16;
+        samples_per_ms_ = 8;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bwb:
+        bytes_per_ms_ = 32;
+        samples_per_ms_ = 16;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bswb32kHz:
+        bytes_per_ms_ = 64;
+        samples_per_ms_ = 32;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bswb48kHz:
+        bytes_per_ms_ = 96;
+        samples_per_ms_ = 48;
+        break;
+      case NetEqDecoder::kDecoderPCM16B_2ch:
+        bytes_per_ms_ = 2 * 16;
+        samples_per_ms_ = 8;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bwb_2ch:
+        bytes_per_ms_ = 2 * 32;
+        samples_per_ms_ = 16;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch:
+        bytes_per_ms_ = 2 * 64;
+        samples_per_ms_ = 32;
+        break;
+      case NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch:
+        bytes_per_ms_ = 2 * 96;
+        samples_per_ms_ = 48;
+        break;
+      case NetEqDecoder::kDecoderPCM16B_5ch:
+        bytes_per_ms_ = 5 * 16;
+        samples_per_ms_ = 8;
+        break;
+      default:
+        RTC_DCHECK_NOTREACHED();
+        break;
+    }
+  }
+  size_t bytes_per_ms_;
+  int samples_per_ms_;
+  NetEqDecoder decoder_type_;
+};
+
+// Test splitting sample-based payloads.
+TEST_P(SplitBySamplesTest, PayloadSizes) {
+  constexpr uint32_t kBaseTimestamp = 0x12345678;
+  struct ExpectedSplit {
+    size_t payload_size_ms;
+    size_t num_frames;
+    // For simplicity. We only expect up to two packets per split.
+    size_t frame_sizes[2];
+  };
+  // The payloads are expected to be split as follows:
+  // 10 ms -> 10 ms
+  // 20 ms -> 20 ms
+  // 30 ms -> 30 ms
+  // 40 ms -> 20 + 20 ms
+  // 50 ms -> 25 + 25 ms
+  // 60 ms -> 30 + 30 ms
+  ExpectedSplit expected_splits[] = {{10, 1, {10}},     {20, 1, {20}},
+                                     {30, 1, {30}},     {40, 2, {20, 20}},
+                                     {50, 2, {25, 25}}, {60, 2, {30, 30}}};
+
+  for (const auto& expected_split : expected_splits) {
+    // The payload values are set to steadily increase (modulo 256), so that the
+    // resulting frames can be checked and we can be reasonably certain no
+    // sample was missed or repeated.
+    const auto generate_payload = [](size_t num_bytes) {
+      rtc::Buffer payload(num_bytes);
+      uint8_t value = 0;
+      // Allow wrap-around of value in counter below.
+      for (size_t i = 0; i != payload.size(); ++i, ++value) {
+        payload[i] = value;
+      }
+      return payload;
+    };
+
+    const auto results = LegacyEncodedAudioFrame::SplitBySamples(
+        nullptr,
+        generate_payload(expected_split.payload_size_ms * bytes_per_ms_),
+        kBaseTimestamp, bytes_per_ms_, samples_per_ms_);
+
+    EXPECT_EQ(expected_split.num_frames, results.size());
+    uint32_t expected_timestamp = kBaseTimestamp;
+    uint8_t value = 0;
+    for (size_t i = 0; i != expected_split.num_frames; ++i) {
+      const auto& result = results[i];
+      const LegacyEncodedAudioFrame* frame =
+          static_cast<const LegacyEncodedAudioFrame*>(result.frame.get());
+      const size_t length_bytes = expected_split.frame_sizes[i] * bytes_per_ms_;
+      EXPECT_EQ(length_bytes, frame->payload().size());
+      EXPECT_EQ(expected_timestamp, result.timestamp);
+      const rtc::Buffer& payload = frame->payload();
+      // Allow wrap-around of value in counter below.
+      for (size_t i = 0; i != payload.size(); ++i, ++value) {
+        ASSERT_EQ(value, payload[i]);
+      }
+
+      expected_timestamp += rtc::checked_cast<uint32_t>(
+          expected_split.frame_sizes[i] * samples_per_ms_);
+    }
+  }
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    LegacyEncodedAudioFrame,
+    SplitBySamplesTest,
+    ::testing::Values(NetEqDecoder::kDecoderPCMu,
+                      NetEqDecoder::kDecoderPCMa,
+                      NetEqDecoder::kDecoderPCMu_2ch,
+                      NetEqDecoder::kDecoderPCMa_2ch,
+                      NetEqDecoder::kDecoderG722,
+                      NetEqDecoder::kDecoderPCM16B,
+                      NetEqDecoder::kDecoderPCM16Bwb,
+                      NetEqDecoder::kDecoderPCM16Bswb32kHz,
+                      NetEqDecoder::kDecoderPCM16Bswb48kHz,
+                      NetEqDecoder::kDecoderPCM16B_2ch,
+                      NetEqDecoder::kDecoderPCM16Bwb_2ch,
+                      NetEqDecoder::kDecoderPCM16Bswb32kHz_2ch,
+                      NetEqDecoder::kDecoderPCM16Bswb48kHz_2ch,
+                      NetEqDecoder::kDecoderPCM16B_5ch));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/DEPS b/third_party/libwebrtc/modules/audio_coding/codecs/opus/DEPS
new file mode 100644
index 0000000000..c2530726ad
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/DEPS
@@ -0,0 +1,5 @@
+specific_include_rules = {
+  "opus_inst\.h": [
+    "+third_party/opus",
+  ],
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc
new file mode 100644
index 0000000000..03c02186d0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc
@@ -0,0 +1,52 @@
+/*
+ *  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 "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
+                                               absl::string_view param) {
+  auto it = format.parameters.find(std::string(param));
+  if (it == format.parameters.end())
+    return absl::nullopt;
+
+  return it->second;
+}
+
+// Parses a comma-separated string "1,2,0,6" into a std::vector<unsigned char>.
+template <>
+absl::optional<std::vector<unsigned char>> GetFormatParameter(
+    const SdpAudioFormat& format,
+    absl::string_view param) {
+  std::vector<unsigned char> result;
+  const std::string comma_separated_list =
+      GetFormatParameter(format, param).value_or("");
+  size_t pos = 0;
+  while (pos < comma_separated_list.size()) {
+    const size_t next_comma = comma_separated_list.find(',', pos);
+    const size_t distance_to_next_comma = next_comma == std::string::npos
+                                              ? std::string::npos
+                                              : (next_comma - pos);
+    auto substring_with_number =
+        comma_separated_list.substr(pos, distance_to_next_comma);
+    auto conv = rtc::StringToNumber<int>(substring_with_number);
+    if (!conv.has_value()) {
+      return absl::nullopt;
+    }
+    result.push_back(*conv);
+    pos += substring_with_number.size() + 1;
+  }
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
new file mode 100644
index 0000000000..5ebb51b577
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
@@ -0,0 +1,89 @@
+/*
+ *  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 MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
+                                               absl::string_view param);
+
+template <typename T>
+absl::optional<T> GetFormatParameter(const SdpAudioFormat& format,
+                                     absl::string_view param) {
+  return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or(""));
+}
+
+template <>
+absl::optional<std::vector<unsigned char>> GetFormatParameter(
+    const SdpAudioFormat& format,
+    absl::string_view param);
+
+class OpusFrame : public AudioDecoder::EncodedAudioFrame {
+ public:
+  OpusFrame(AudioDecoder* decoder,
+            rtc::Buffer&& payload,
+            bool is_primary_payload)
+      : decoder_(decoder),
+        payload_(std::move(payload)),
+        is_primary_payload_(is_primary_payload) {}
+
+  size_t Duration() const override {
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    } else {
+      ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size());
+    }
+    return (ret < 0) ? 0 : static_cast<size_t>(ret);
+  }
+
+  bool IsDtxPacket() const override { return payload_.size() <= 2; }
+
+  absl::optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->Decode(
+          payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+          decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    } else {
+      ret = decoder_->DecodeRedundant(
+          payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+          decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    }
+
+    if (ret < 0)
+      return absl::nullopt;
+
+    return DecodeResult{static_cast<size_t>(ret), speech_type};
+  }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+  const bool is_primary_payload_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
new file mode 100644
index 0000000000..285ea89959
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
@@ -0,0 +1,182 @@
+/*
+ *  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 "modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h"
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+std::unique_ptr<AudioDecoderMultiChannelOpusImpl>
+AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(
+    AudioDecoderMultiChannelOpusConfig config) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  // Fill the pointer with a working decoder through the C interface. This
+  // allocates memory.
+  OpusDecInst* dec_state = nullptr;
+  const int error = WebRtcOpus_MultistreamDecoderCreate(
+      &dec_state, config.num_channels, config.num_streams,
+      config.coupled_streams, config.channel_mapping.data());
+  if (error != 0) {
+    return nullptr;
+  }
+
+  // Pass the ownership to DecoderImpl. Not using 'make_unique' because the
+  // c-tor is private.
+  return std::unique_ptr<AudioDecoderMultiChannelOpusImpl>(
+      new AudioDecoderMultiChannelOpusImpl(dec_state, config));
+}
+
+AudioDecoderMultiChannelOpusImpl::AudioDecoderMultiChannelOpusImpl(
+    OpusDecInst* dec_state,
+    AudioDecoderMultiChannelOpusConfig config)
+    : dec_state_(dec_state), config_(config) {
+  RTC_DCHECK(dec_state);
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderMultiChannelOpusImpl::~AudioDecoderMultiChannelOpusImpl() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+absl::optional<AudioDecoderMultiChannelOpusConfig>
+AudioDecoderMultiChannelOpusImpl::SdpToConfig(const SdpAudioFormat& format) {
+  AudioDecoderMultiChannelOpusConfig config;
+  config.num_channels = format.num_channels;
+  auto num_streams = GetFormatParameter<int>(format, "num_streams");
+  if (!num_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.num_streams = *num_streams;
+
+  auto coupled_streams = GetFormatParameter<int>(format, "coupled_streams");
+  if (!coupled_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.coupled_streams = *coupled_streams;
+
+  auto channel_mapping =
+      GetFormatParameter<std::vector<unsigned char>>(format, "channel_mapping");
+  if (!channel_mapping.has_value()) {
+    return absl::nullopt;
+  }
+  config.channel_mapping = *channel_mapping;
+  if (!config.IsOk()) {
+    return absl::nullopt;
+  }
+  return config;
+}
+
+std::vector<AudioDecoder::ParseResult>
+AudioDecoderMultiChannelOpusImpl::ParsePayload(rtc::Buffer&& payload,
+                                               uint32_t timestamp) {
+  std::vector<ParseResult> results;
+
+  if (PacketHasFec(payload.data(), payload.size())) {
+    const int duration =
+        PacketDurationRedundant(payload.data(), payload.size());
+    RTC_DCHECK_GE(duration, 0);
+    rtc::Buffer payload_copy(payload.data(), payload.size());
+    std::unique_ptr<EncodedAudioFrame> fec_frame(
+        new OpusFrame(this, std::move(payload_copy), false));
+    results.emplace_back(timestamp - duration, 1, std::move(fec_frame));
+  }
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OpusFrame(this, std::move(payload), true));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int AudioDecoderMultiChannelOpusImpl::DecodeInternal(const uint8_t* encoded,
+                                                     size_t encoded_len,
+                                                     int sample_rate_hz,
+                                                     int16_t* decoded,
+                                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(
+        config_.num_channels);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderMultiChannelOpusImpl::DecodeRedundantInternal(
+    const uint8_t* encoded,
+    size_t encoded_len,
+    int sample_rate_hz,
+    int16_t* decoded,
+    SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(
+        config_.num_channels);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderMultiChannelOpusImpl::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderMultiChannelOpusImpl::PacketDuration(const uint8_t* encoded,
+                                                     size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderMultiChannelOpusImpl::PacketDurationRedundant(
+    const uint8_t* encoded,
+    size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len, 48000);
+}
+
+bool AudioDecoderMultiChannelOpusImpl::PacketHasFec(const uint8_t* encoded,
+                                                    size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+int AudioDecoderMultiChannelOpusImpl::SampleRateHz() const {
+  return 48000;
+}
+
+size_t AudioDecoderMultiChannelOpusImpl::Channels() const {
+  return config_.num_channels;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
new file mode 100644
index 0000000000..2ff47a8a53
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
@@ -0,0 +1,74 @@
+/*
+ *  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 MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoderMultiChannelOpusImpl final : public AudioDecoder {
+ public:
+  static std::unique_ptr<AudioDecoderMultiChannelOpusImpl> MakeAudioDecoder(
+      AudioDecoderMultiChannelOpusConfig config);
+
+  ~AudioDecoderMultiChannelOpusImpl() override;
+
+  AudioDecoderMultiChannelOpusImpl(const AudioDecoderMultiChannelOpusImpl&) =
+      delete;
+  AudioDecoderMultiChannelOpusImpl& operator=(
+      const AudioDecoderMultiChannelOpusImpl&) = delete;
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+  static absl::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  int DecodeRedundantInternal(const uint8_t* encoded,
+                              size_t encoded_len,
+                              int sample_rate_hz,
+                              int16_t* decoded,
+                              SpeechType* speech_type) override;
+
+ private:
+  AudioDecoderMultiChannelOpusImpl(OpusDecInst* dec_state,
+                                   AudioDecoderMultiChannelOpusConfig config);
+
+  OpusDecInst* dec_state_;
+  const AudioDecoderMultiChannelOpusConfig config_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc
new file mode 100644
index 0000000000..57e2107f3c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+using ::testing::NiceMock;
+using ::testing::Return;
+
+TEST(AudioDecoderMultiOpusTest, GetFormatParameter) {
+  const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                  {{"channel_mapping", "0,1,2,3"},
+                                   {"coupled_streams", "2"},
+                                   {"num_streams", "2"}});
+
+  EXPECT_EQ(GetFormatParameter(sdp_format, "channel_mapping"),
+            absl::optional<std::string>("0,1,2,3"));
+
+  EXPECT_EQ(GetFormatParameter<int>(sdp_format, "coupled_streams"),
+            absl::optional<int>(2));
+
+  EXPECT_EQ(GetFormatParameter(sdp_format, "missing"), absl::nullopt);
+
+  EXPECT_EQ(GetFormatParameter<int>(sdp_format, "channel_mapping"),
+            absl::nullopt);
+}
+
+TEST(AudioDecoderMultiOpusTest, InvalidChannelMappings) {
+  {
+    // Can't use channel 3 if there are only 2 channels.
+    const SdpAudioFormat sdp_format("multiopus", 48000, 2,
+                                    {{"channel_mapping", "3,0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+  {
+    // The mapping is too long. There are only 5 channels, but 6 elements in the
+    // mapping.
+    const SdpAudioFormat sdp_format("multiopus", 48000, 5,
+                                    {{"channel_mapping", "0,1,2,3,4,5"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+  {
+    // The mapping doesn't parse correctly.
+    const SdpAudioFormat sdp_format(
+        "multiopus", 48000, 5,
+        {{"channel_mapping", "0,1,two,3,4"}, {"coupled_streams", "0"}});
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+}
+
+TEST(AudioDecoderMultiOpusTest, ValidSdpToConfigProducesCorrectConfig) {
+  const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                  {{"channel_mapping", "3,1,2,0"},
+                                   {"coupled_streams", "2"},
+                                   {"num_streams", "2"}});
+
+  const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+      AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+  ASSERT_TRUE(decoder_config.has_value());
+  EXPECT_TRUE(decoder_config->IsOk());
+  EXPECT_EQ(decoder_config->coupled_streams, 2);
+  EXPECT_THAT(decoder_config->channel_mapping,
+              ::testing::ContainerEq(std::vector<unsigned char>({3, 1, 2, 0})));
+}
+
+TEST(AudioDecoderMultiOpusTest, InvalidSdpToConfigDoesNotProduceConfig) {
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                    {{"channel_mapping", "0,1,2,3"},
+                                     {"coupled_stream", "2"},
+                                     {"num_streams", "2"}});
+
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                    {{"channel_mapping", "0,1,2 3"},
+                                     {"coupled_streams", "2"},
+                                     {"num_streams", "2"}});
+
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+}
+
+TEST(AudioDecoderMultiOpusTest, CodecsCanBeCreated) {
+  const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                  {{"channel_mapping", "0,1,2,3"},
+                                   {"coupled_streams", "2"},
+                                   {"num_streams", "2"}});
+
+  const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+      AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+  ASSERT_TRUE(decoder_config.has_value());
+
+  const std::unique_ptr<AudioDecoder> opus_decoder =
+      AudioDecoderMultiChannelOpus::MakeAudioDecoder(*decoder_config);
+
+  EXPECT_TRUE(opus_decoder);
+}
+
+TEST(AudioDecoderMultiOpusTest, AdvertisedCodecsCanBeCreated) {
+  std::vector<AudioCodecSpec> specs;
+  AudioDecoderMultiChannelOpus::AppendSupportedDecoders(&specs);
+
+  EXPECT_FALSE(specs.empty());
+
+  for (const AudioCodecSpec& spec : specs) {
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(spec.format);
+    ASSERT_TRUE(decoder_config.has_value());
+
+    const std::unique_ptr<AudioDecoder> opus_decoder =
+        AudioDecoderMultiChannelOpus::MakeAudioDecoder(*decoder_config);
+
+    EXPECT_TRUE(opus_decoder);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
new file mode 100644
index 0000000000..cff9685548
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels,
+                                           int sample_rate_hz)
+    : channels_{num_channels}, sample_rate_hz_{sample_rate_hz} {
+  RTC_DCHECK(num_channels == 1 || num_channels == 2);
+  RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000);
+  const int error =
+      WebRtcOpus_DecoderCreate(&dec_state_, channels_, sample_rate_hz_);
+  RTC_DCHECK(error == 0);
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderOpusImpl::~AudioDecoderOpusImpl() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderOpusImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+
+  if (PacketHasFec(payload.data(), payload.size())) {
+    const int duration =
+        PacketDurationRedundant(payload.data(), payload.size());
+    RTC_DCHECK_GE(duration, 0);
+    rtc::Buffer payload_copy(payload.data(), payload.size());
+    std::unique_ptr<EncodedAudioFrame> fec_frame(
+        new OpusFrame(this, std::move(payload_copy), false));
+    results.emplace_back(timestamp - duration, 1, std::move(fec_frame));
+  }
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OpusFrame(this, std::move(payload), true));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded,
+                                                  size_t encoded_len,
+                                                  int sample_rate_hz,
+                                                  int16_t* decoded,
+                                                  SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderOpusImpl::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderOpusImpl::PacketDuration(const uint8_t* encoded,
+                                         size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded,
+                                                  size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len, sample_rate_hz_);
+}
+
+bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded,
+                                        size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+int AudioDecoderOpusImpl::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
+size_t AudioDecoderOpusImpl::Channels() const {
+  return channels_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
new file mode 100644
index 0000000000..e8fd0440bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoderOpusImpl final : public AudioDecoder {
+ public:
+  explicit AudioDecoderOpusImpl(size_t num_channels,
+                                int sample_rate_hz = 48000);
+  ~AudioDecoderOpusImpl() override;
+
+  AudioDecoderOpusImpl(const AudioDecoderOpusImpl&) = delete;
+  AudioDecoderOpusImpl& operator=(const AudioDecoderOpusImpl&) = delete;
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  int DecodeRedundantInternal(const uint8_t* encoded,
+                              size_t encoded_len,
+                              int sample_rate_hz,
+                              int16_t* decoded,
+                              SpeechType* speech_type) override;
+
+ private:
+  OpusDecInst* dec_state_;
+  const size_t channels_;
+  const int sample_rate_hz_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
new file mode 100644
index 0000000000..38a11c123d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
@@ -0,0 +1,366 @@
+/*
+ *  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.
+ */
+
+/*
+ * LEFT TO DO:
+ * - WRITE TESTS for the stuff in this  file.
+ * - Check the creation, maybe make it safer by returning an empty optional or
+ *   unique_ptr. --- It looks OK, but RecreateEncoderInstance can perhaps crash
+ *   on a valid config. Can run it in the fuzzer for some time. Should prbl also
+ *   fuzz the config.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h"
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+namespace {
+
+// Recommended bitrates for one channel:
+// 8-12 kb/s for NB speech,
+// 16-20 kb/s for WB speech,
+// 28-40 kb/s for FB speech,
+// 48-64 kb/s for FB mono music, and
+// 64-128 kb/s for FB stereo music.
+// The current implementation multiplies these values by the number of channels.
+constexpr int kOpusBitrateNbBps = 12000;
+constexpr int kOpusBitrateWbBps = 20000;
+constexpr int kOpusBitrateFbBps = 32000;
+
+constexpr int kDefaultMaxPlaybackRate = 48000;
+// These two lists must be sorted from low to high
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60, 120};
+#else
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60};
+#endif
+
+int GetBitrateBps(const AudioEncoderMultiChannelOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return config.bitrate_bps;
+}
+int GetMaxPlaybackRate(const SdpAudioFormat& format) {
+  const auto param = GetFormatParameter<int>(format, "maxplaybackrate");
+  if (param && *param >= 8000) {
+    return std::min(*param, kDefaultMaxPlaybackRate);
+  }
+  return kDefaultMaxPlaybackRate;
+}
+
+int GetFrameSizeMs(const SdpAudioFormat& format) {
+  const auto ptime = GetFormatParameter<int>(format, "ptime");
+  if (ptime.has_value()) {
+    // Pick the next highest supported frame length from
+    // kOpusSupportedFrameLengths.
+    for (const int supported_frame_length : kOpusSupportedFrameLengths) {
+      if (supported_frame_length >= *ptime) {
+        return supported_frame_length;
+      }
+    }
+    // If none was found, return the largest supported frame length.
+    return *(std::end(kOpusSupportedFrameLengths) - 1);
+  }
+
+  return AudioEncoderOpusConfig::kDefaultFrameSizeMs;
+}
+
+int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
+  const int bitrate = [&] {
+    if (max_playback_rate <= 8000) {
+      return kOpusBitrateNbBps * rtc::dchecked_cast<int>(num_channels);
+    } else if (max_playback_rate <= 16000) {
+      return kOpusBitrateWbBps * rtc::dchecked_cast<int>(num_channels);
+    } else {
+      return kOpusBitrateFbBps * rtc::dchecked_cast<int>(num_channels);
+    }
+  }();
+  RTC_DCHECK_GE(bitrate, AudioEncoderMultiChannelOpusConfig::kMinBitrateBps);
+  return bitrate;
+}
+
+// Get the maxaveragebitrate parameter in string-form, so we can properly figure
+// out how invalid it is and accurately log invalid values.
+int CalculateBitrate(int max_playback_rate_hz,
+                     size_t num_channels,
+                     absl::optional<std::string> bitrate_param) {
+  const int default_bitrate =
+      CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
+
+  if (bitrate_param) {
+    const auto bitrate = rtc::StringToNumber<int>(*bitrate_param);
+    if (bitrate) {
+      const int chosen_bitrate =
+          std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                   std::min(*bitrate, AudioEncoderOpusConfig::kMaxBitrateBps));
+      if (bitrate != chosen_bitrate) {
+        RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate
+                            << " clamped to " << chosen_bitrate;
+      }
+      return chosen_bitrate;
+    }
+    RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate \"" << *bitrate_param
+                        << "\" replaced by default bitrate " << default_bitrate;
+  }
+
+  return default_bitrate;
+}
+
+}  // namespace
+
+std::unique_ptr<AudioEncoder>
+AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config,
+    int payload_type) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderMultiChannelOpusImpl>(config,
+                                                            payload_type);
+}
+
+AudioEncoderMultiChannelOpusImpl::AudioEncoderMultiChannelOpusImpl(
+    const AudioEncoderMultiChannelOpusConfig& config,
+    int payload_type)
+    : payload_type_(payload_type), inst_(nullptr) {
+  RTC_DCHECK(0 <= payload_type && payload_type <= 127);
+
+  RTC_CHECK(RecreateEncoderInstance(config));
+}
+
+AudioEncoderMultiChannelOpusImpl::~AudioEncoderMultiChannelOpusImpl() {
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+}
+
+size_t AudioEncoderMultiChannelOpusImpl::SufficientOutputBufferSize() const {
+  // Calculate the number of bytes we expect the encoder to produce,
+  // then multiply by two to give a wide margin for error.
+  const size_t bytes_per_millisecond =
+      static_cast<size_t>(GetBitrateBps(config_) / (1000 * 8) + 1);
+  const size_t approx_encoded_bytes =
+      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
+  return 2 * approx_encoded_bytes;
+}
+
+void AudioEncoderMultiChannelOpusImpl::Reset() {
+  RTC_CHECK(RecreateEncoderInstance(config_));
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderMultiChannelOpusImpl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(config_.frame_size_ms),
+           TimeDelta::Millis(config_.frame_size_ms)}};
+}
+
+// If the given config is OK, recreate the Opus encoder instance with those
+// settings, save the config, and return true. Otherwise, do nothing and return
+// false.
+bool AudioEncoderMultiChannelOpusImpl::RecreateEncoderInstance(
+    const AudioEncoderMultiChannelOpusConfig& config) {
+  if (!config.IsOk())
+    return false;
+  config_ = config;
+  if (inst_)
+    RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+  input_buffer_.clear();
+  input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_MultistreamEncoderCreate(
+             &inst_, config.num_channels,
+             config.application ==
+                     AudioEncoderMultiChannelOpusConfig::ApplicationMode::kVoip
+                 ? 0
+                 : 1,
+             config.num_streams, config.coupled_streams,
+             config.channel_mapping.data()));
+  const int bitrate = GetBitrateBps(config);
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, bitrate));
+  RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
+  if (config.fec_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus enable FEC";
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus disable FEC";
+  }
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
+  RTC_LOG(LS_VERBOSE) << "Set Opus playback rate to "
+                      << config.max_playback_rate_hz << " hz.";
+
+  // Use the DEFAULT complexity.
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_SetComplexity(inst_, AudioEncoderOpusConfig().complexity));
+  RTC_LOG(LS_VERBOSE) << "Set Opus coding complexity to "
+                      << AudioEncoderOpusConfig().complexity;
+
+  if (config.dtx_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus enable DTX";
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus disable DTX";
+  }
+
+  if (config.cbr_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableCbr(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus enable CBR";
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableCbr(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus disable CBR";
+  }
+  num_channels_to_encode_ = NumChannels();
+  next_frame_length_ms_ = config_.frame_size_ms;
+  RTC_LOG(LS_VERBOSE) << "Set Opus frame length to " << config_.frame_size_ms
+                      << " ms";
+  return true;
+}
+
+absl::optional<AudioEncoderMultiChannelOpusConfig>
+AudioEncoderMultiChannelOpusImpl::SdpToConfig(const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name, "multiopus") ||
+      format.clockrate_hz != 48000) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderMultiChannelOpusConfig config;
+  config.num_channels = format.num_channels;
+  config.frame_size_ms = GetFrameSizeMs(format);
+  config.max_playback_rate_hz = GetMaxPlaybackRate(format);
+  config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
+  config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
+  config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
+  config.bitrate_bps =
+      CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
+                       GetFormatParameter(format, "maxaveragebitrate"));
+  config.application =
+      config.num_channels == 1
+          ? AudioEncoderMultiChannelOpusConfig::ApplicationMode::kVoip
+          : AudioEncoderMultiChannelOpusConfig::ApplicationMode::kAudio;
+
+  config.supported_frame_lengths_ms.clear();
+  std::copy(std::begin(kOpusSupportedFrameLengths),
+            std::end(kOpusSupportedFrameLengths),
+            std::back_inserter(config.supported_frame_lengths_ms));
+
+  auto num_streams = GetFormatParameter<int>(format, "num_streams");
+  if (!num_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.num_streams = *num_streams;
+
+  auto coupled_streams = GetFormatParameter<int>(format, "coupled_streams");
+  if (!coupled_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.coupled_streams = *coupled_streams;
+
+  auto channel_mapping =
+      GetFormatParameter<std::vector<unsigned char>>(format, "channel_mapping");
+  if (!channel_mapping.has_value()) {
+    return absl::nullopt;
+  }
+  config.channel_mapping = *channel_mapping;
+
+  if (!config.IsOk()) {
+    return absl::nullopt;
+  }
+  return config;
+}
+
+AudioCodecInfo AudioEncoderMultiChannelOpusImpl::QueryAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  AudioCodecInfo info(48000, config.num_channels, config.bitrate_bps,
+                      AudioEncoderOpusConfig::kMinBitrateBps,
+                      AudioEncoderOpusConfig::kMaxBitrateBps);
+  info.allow_comfort_noise = false;
+  info.supports_network_adaption = false;
+  return info;
+}
+
+size_t AudioEncoderMultiChannelOpusImpl::Num10msFramesPerPacket() const {
+  return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
+}
+size_t AudioEncoderMultiChannelOpusImpl::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(48000, 100) * config_.num_channels;
+}
+int AudioEncoderMultiChannelOpusImpl::SampleRateHz() const {
+  return 48000;
+}
+size_t AudioEncoderMultiChannelOpusImpl::NumChannels() const {
+  return config_.num_channels;
+}
+size_t AudioEncoderMultiChannelOpusImpl::Num10MsFramesInNextPacket() const {
+  return Num10msFramesPerPacket();
+}
+size_t AudioEncoderMultiChannelOpusImpl::Max10MsFramesInAPacket() const {
+  return Num10msFramesPerPacket();
+}
+int AudioEncoderMultiChannelOpusImpl::GetTargetBitrate() const {
+  return GetBitrateBps(config_);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderMultiChannelOpusImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (input_buffer_.empty())
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
+  if (input_buffer_.size() <
+      (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(input_buffer_.size(),
+               Num10msFramesPerPacket() * SamplesPer10msFrame());
+
+  const size_t max_encoded_bytes = SufficientOutputBufferSize();
+  EncodedInfo info;
+  info.encoded_bytes = encoded->AppendData(
+      max_encoded_bytes, [&](rtc::ArrayView<uint8_t> encoded) {
+        int status = WebRtcOpus_Encode(
+            inst_, &input_buffer_[0],
+            rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
+            rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded.data());
+
+        RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
+
+        return static_cast<size_t>(status);
+      });
+  input_buffer_.clear();
+
+  // Will use new packet size for next encoding.
+  config_.frame_size_ms = next_frame_length_ms_;
+
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.send_even_if_empty = true;  // Allows Opus to send empty packets.
+
+  info.speech = true;
+  info.encoder_type = CodecType::kOther;
+
+  return info;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
new file mode 100644
index 0000000000..8a7210515c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
@@ -0,0 +1,92 @@
+/*
+ *  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 MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class AudioEncoderMultiChannelOpusImpl final : public AudioEncoder {
+ public:
+  AudioEncoderMultiChannelOpusImpl(
+      const AudioEncoderMultiChannelOpusConfig& config,
+      int payload_type);
+  ~AudioEncoderMultiChannelOpusImpl() override;
+
+  AudioEncoderMultiChannelOpusImpl(const AudioEncoderMultiChannelOpusImpl&) =
+      delete;
+  AudioEncoderMultiChannelOpusImpl& operator=(
+      const AudioEncoderMultiChannelOpusImpl&) = delete;
+
+  // Static interface for use by BuiltinAudioEncoderFactory.
+  static constexpr const char* GetPayloadName() { return "multiopus"; }
+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format);
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  static absl::optional<AudioEncoderMultiChannelOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+  static AudioCodecInfo QueryAudioEncoder(
+      const AudioEncoderMultiChannelOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderMultiChannelOpusConfig&,
+      int payload_type);
+
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
+  size_t SufficientOutputBufferSize() const;
+  bool RecreateEncoderInstance(
+      const AudioEncoderMultiChannelOpusConfig& config);
+  void SetFrameLength(int frame_length_ms);
+  void SetNumChannelsToEncode(size_t num_channels_to_encode);
+  void SetProjectedPacketLossRate(float fraction);
+
+  AudioEncoderMultiChannelOpusConfig config_;
+  const int payload_type_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+  size_t num_channels_to_encode_;
+  int next_frame_length_ms_;
+
+  friend struct AudioEncoderMultiChannelOpus;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc
new file mode 100644
index 0000000000..92f6f2c169
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
+
+#include "test/gmock.h"
+
+namespace webrtc {
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace {
+constexpr int kOpusPayloadType = 120;
+}  // namespace
+
+TEST(AudioEncoderMultiOpusTest, CheckConfigValidity) {
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 2,
+                                    {{"channel_mapping", "3,0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // Maps input channel 0 to coded channel 3, which doesn't exist.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 2,
+                                    {{"channel_mapping", "0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // The mapping is too short.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,0,0"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "1"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // Coded channel 0 comes from both input channels 0, 1 and 2.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,255,255"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "1"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+    ASSERT_TRUE(encoder_config.has_value());
+
+    // This is fine, because channels 1, 2 are set to be ignored.
+    EXPECT_TRUE(encoder_config->IsOk());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,255,255"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // This is NOT fine, because channels nothing says how coded channel 1
+    // should be coded.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+}
+
+TEST(AudioEncoderMultiOpusTest, ConfigValuesAreParsedCorrectly) {
+  SdpAudioFormat sdp_format({"multiopus",
+                             48000,
+                             6,
+                             {{"minptime", "10"},
+                              {"useinbandfec", "1"},
+                              {"channel_mapping", "0,4,1,2,3,5"},
+                              {"num_streams", "4"},
+                              {"coupled_streams", "2"}}});
+  const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+      AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+  ASSERT_TRUE(encoder_config.has_value());
+
+  EXPECT_EQ(encoder_config->coupled_streams, 2);
+  EXPECT_EQ(encoder_config->num_streams, 4);
+  EXPECT_THAT(
+      encoder_config->channel_mapping,
+      testing::ContainerEq(std::vector<unsigned char>({0, 4, 1, 2, 3, 5})));
+}
+
+TEST(AudioEncoderMultiOpusTest, CreateFromValidConfig) {
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,255,255"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+    ASSERT_FALSE(encoder_config.has_value());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "1,255,0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "1"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+    ASSERT_TRUE(encoder_config.has_value());
+
+    EXPECT_THAT(encoder_config->channel_mapping,
+                testing::ContainerEq(std::vector<unsigned char>({1, 255, 0})));
+
+    EXPECT_TRUE(encoder_config->IsOk());
+
+    const std::unique_ptr<AudioEncoder> opus_encoder =
+        AudioEncoderMultiChannelOpus::MakeAudioEncoder(*encoder_config,
+                                                       kOpusPayloadType);
+
+    // Creating an encoder from a valid config should work.
+    EXPECT_TRUE(opus_encoder);
+  }
+}
+
+TEST(AudioEncoderMultiOpusTest, AdvertisedCodecsCanBeCreated) {
+  std::vector<AudioCodecSpec> specs;
+  AudioEncoderMultiChannelOpus::AppendSupportedEncoders(&specs);
+
+  EXPECT_FALSE(specs.empty());
+
+  for (const AudioCodecSpec& spec : specs) {
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(spec.format);
+    ASSERT_TRUE(encoder_config.has_value());
+
+    const std::unique_ptr<AudioEncoder> opus_encoder =
+        AudioEncoderMultiChannelOpus::MakeAudioEncoder(*encoder_config,
+                                                       kOpusPayloadType);
+
+    EXPECT_TRUE(opus_encoder);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
new file mode 100644
index 0000000000..17e0e33b1d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -0,0 +1,824 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+
+#include <algorithm>
+#include <iterator>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/exp_filter.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/string_to_number.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+// Codec parameters for Opus.
+// draft-spittka-payload-rtp-opus-03
+
+// Recommended bitrates:
+// 8-12 kb/s for NB speech,
+// 16-20 kb/s for WB speech,
+// 28-40 kb/s for FB speech,
+// 48-64 kb/s for FB mono music, and
+// 64-128 kb/s for FB stereo music.
+// The current implementation applies the following values to mono signals,
+// and multiplies them by 2 for stereo.
+constexpr int kOpusBitrateNbBps = 12000;
+constexpr int kOpusBitrateWbBps = 20000;
+constexpr int kOpusBitrateFbBps = 32000;
+
+constexpr int kRtpTimestampRateHz = 48000;
+constexpr int kDefaultMaxPlaybackRate = 48000;
+
+// These two lists must be sorted from low to high
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+constexpr int kANASupportedFrameLengths[] = {20, 40, 60, 120};
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60, 120};
+#else
+constexpr int kANASupportedFrameLengths[] = {20, 40, 60};
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60};
+#endif
+
+// PacketLossFractionSmoother uses an exponential filter with a time constant
+// of -1.0 / ln(0.9999) = 10000 ms.
+constexpr float kAlphaForPacketLossFractionSmoother = 0.9999f;
+constexpr float kMaxPacketLossFraction = 0.2f;
+
+int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
+  const int bitrate = [&] {
+    if (max_playback_rate <= 8000) {
+      return kOpusBitrateNbBps * rtc::dchecked_cast<int>(num_channels);
+    } else if (max_playback_rate <= 16000) {
+      return kOpusBitrateWbBps * rtc::dchecked_cast<int>(num_channels);
+    } else {
+      return kOpusBitrateFbBps * rtc::dchecked_cast<int>(num_channels);
+    }
+  }();
+  RTC_DCHECK_GE(bitrate, AudioEncoderOpusConfig::kMinBitrateBps);
+  RTC_DCHECK_LE(bitrate, AudioEncoderOpusConfig::kMaxBitrateBps);
+  return bitrate;
+}
+
+// Get the maxaveragebitrate parameter in string-form, so we can properly figure
+// out how invalid it is and accurately log invalid values.
+int CalculateBitrate(int max_playback_rate_hz,
+                     size_t num_channels,
+                     absl::optional<std::string> bitrate_param) {
+  const int default_bitrate =
+      CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
+
+  if (bitrate_param) {
+    const auto bitrate = rtc::StringToNumber<int>(*bitrate_param);
+    if (bitrate) {
+      const int chosen_bitrate =
+          std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                   std::min(*bitrate, AudioEncoderOpusConfig::kMaxBitrateBps));
+      if (bitrate != chosen_bitrate) {
+        RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate
+                            << " clamped to " << chosen_bitrate;
+      }
+      return chosen_bitrate;
+    }
+    RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate \"" << *bitrate_param
+                        << "\" replaced by default bitrate " << default_bitrate;
+  }
+
+  return default_bitrate;
+}
+
+int GetChannelCount(const SdpAudioFormat& format) {
+  const auto param = GetFormatParameter(format, "stereo");
+  if (param == "1") {
+    return 2;
+  } else {
+    return 1;
+  }
+}
+
+int GetMaxPlaybackRate(const SdpAudioFormat& format) {
+  const auto param = GetFormatParameter<int>(format, "maxplaybackrate");
+  if (param && *param >= 8000) {
+    return std::min(*param, kDefaultMaxPlaybackRate);
+  }
+  return kDefaultMaxPlaybackRate;
+}
+
+int GetFrameSizeMs(const SdpAudioFormat& format) {
+  const auto ptime = GetFormatParameter<int>(format, "ptime");
+  if (ptime) {
+    // Pick the next highest supported frame length from
+    // kOpusSupportedFrameLengths.
+    for (const int supported_frame_length : kOpusSupportedFrameLengths) {
+      if (supported_frame_length >= *ptime) {
+        return supported_frame_length;
+      }
+    }
+    // If none was found, return the largest supported frame length.
+    return *(std::end(kOpusSupportedFrameLengths) - 1);
+  }
+
+  return AudioEncoderOpusConfig::kDefaultFrameSizeMs;
+}
+
+void FindSupportedFrameLengths(int min_frame_length_ms,
+                               int max_frame_length_ms,
+                               std::vector<int>* out) {
+  out->clear();
+  std::copy_if(std::begin(kANASupportedFrameLengths),
+               std::end(kANASupportedFrameLengths), std::back_inserter(*out),
+               [&](int frame_length_ms) {
+                 return frame_length_ms >= min_frame_length_ms &&
+                        frame_length_ms <= max_frame_length_ms;
+               });
+  RTC_DCHECK(std::is_sorted(out->begin(), out->end()));
+}
+
+int GetBitrateBps(const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return *config.bitrate_bps;
+}
+
+std::vector<float> GetBitrateMultipliers() {
+  constexpr char kBitrateMultipliersName[] =
+      "WebRTC-Audio-OpusBitrateMultipliers";
+  const bool use_bitrate_multipliers =
+      webrtc::field_trial::IsEnabled(kBitrateMultipliersName);
+  if (use_bitrate_multipliers) {
+    const std::string field_trial_string =
+        webrtc::field_trial::FindFullName(kBitrateMultipliersName);
+    std::vector<std::string> pieces;
+    rtc::tokenize(field_trial_string, '-', &pieces);
+    if (pieces.size() < 2 || pieces[0] != "Enabled") {
+      RTC_LOG(LS_WARNING) << "Invalid parameters for "
+                          << kBitrateMultipliersName
+                          << ", not using custom values.";
+      return std::vector<float>();
+    }
+    std::vector<float> multipliers(pieces.size() - 1);
+    for (size_t i = 1; i < pieces.size(); i++) {
+      if (!rtc::FromString(pieces[i], &multipliers[i - 1])) {
+        RTC_LOG(LS_WARNING)
+            << "Invalid parameters for " << kBitrateMultipliersName
+            << ", not using custom values.";
+        return std::vector<float>();
+      }
+    }
+    RTC_LOG(LS_INFO) << "Using custom bitrate multipliers: "
+                     << field_trial_string;
+    return multipliers;
+  }
+  return std::vector<float>();
+}
+
+int GetMultipliedBitrate(int bitrate, const std::vector<float>& multipliers) {
+  // The multipliers are valid from 5 kbps.
+  const size_t bitrate_kbps = static_cast<size_t>(bitrate / 1000);
+  if (bitrate_kbps < 5 || bitrate_kbps >= multipliers.size() + 5) {
+    return bitrate;
+  }
+  return static_cast<int>(multipliers[bitrate_kbps - 5] * bitrate);
+}
+}  // namespace
+
+void AudioEncoderOpusImpl::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"opus",
+                              kRtpTimestampRateHz,
+                              2,
+                              {{"minptime", "10"}, {"useinbandfec", "1"}}};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderOpusImpl::QueryAudioEncoder(
+    const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  AudioCodecInfo info(config.sample_rate_hz, config.num_channels,
+                      *config.bitrate_bps,
+                      AudioEncoderOpusConfig::kMinBitrateBps,
+                      AudioEncoderOpusConfig::kMaxBitrateBps);
+  info.allow_comfort_noise = false;
+  info.supports_network_adaption = true;
+  return info;
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
+    const AudioEncoderOpusConfig& config,
+    int payload_type) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderOpusImpl>(config, payload_type);
+}
+
+absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name, "opus") ||
+      format.clockrate_hz != kRtpTimestampRateHz) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderOpusConfig config;
+  config.num_channels = GetChannelCount(format);
+  config.frame_size_ms = GetFrameSizeMs(format);
+  config.max_playback_rate_hz = GetMaxPlaybackRate(format);
+  config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
+  config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
+  config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
+  config.bitrate_bps =
+      CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
+                       GetFormatParameter(format, "maxaveragebitrate"));
+  config.application = config.num_channels == 1
+                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
+
+  constexpr int kMinANAFrameLength = kANASupportedFrameLengths[0];
+  constexpr int kMaxANAFrameLength =
+      kANASupportedFrameLengths[arraysize(kANASupportedFrameLengths) - 1];
+
+  // For now, minptime and maxptime are only used with ANA. If ptime is outside
+  // of this range, it will get adjusted once ANA takes hold. Ideally, we'd know
+  // if ANA was to be used when setting up the config, and adjust accordingly.
+  const int min_frame_length_ms =
+      GetFormatParameter<int>(format, "minptime").value_or(kMinANAFrameLength);
+  const int max_frame_length_ms =
+      GetFormatParameter<int>(format, "maxptime").value_or(kMaxANAFrameLength);
+
+  FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
+                            &config.supported_frame_lengths_ms);
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+absl::optional<int> AudioEncoderOpusImpl::GetNewComplexity(
+    const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  const int bitrate_bps = GetBitrateBps(config);
+  if (bitrate_bps >= config.complexity_threshold_bps -
+                         config.complexity_threshold_window_bps &&
+      bitrate_bps <= config.complexity_threshold_bps +
+                         config.complexity_threshold_window_bps) {
+    // Within the hysteresis window; make no change.
+    return absl::nullopt;
+  } else {
+    return bitrate_bps <= config.complexity_threshold_bps
+               ? config.low_rate_complexity
+               : config.complexity;
+  }
+}
+
+absl::optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
+    const AudioEncoderOpusConfig& config,
+    OpusEncInst* inst) {
+  constexpr int kMinWidebandBitrate = 8000;
+  constexpr int kMaxNarrowbandBitrate = 9000;
+  constexpr int kAutomaticThreshold = 11000;
+  RTC_DCHECK(config.IsOk());
+  const int bitrate = GetBitrateBps(config);
+  if (bitrate > kAutomaticThreshold) {
+    return absl::optional<int>(OPUS_AUTO);
+  }
+  const int bandwidth = WebRtcOpus_GetBandwidth(inst);
+  RTC_DCHECK_GE(bandwidth, 0);
+  if (bitrate > kMaxNarrowbandBitrate && bandwidth < OPUS_BANDWIDTH_WIDEBAND) {
+    return absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND);
+  } else if (bitrate < kMinWidebandBitrate &&
+             bandwidth > OPUS_BANDWIDTH_NARROWBAND) {
+    return absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND);
+  }
+  return absl::optional<int>();
+}
+
+class AudioEncoderOpusImpl::PacketLossFractionSmoother {
+ public:
+  explicit PacketLossFractionSmoother()
+      : last_sample_time_ms_(rtc::TimeMillis()),
+        smoother_(kAlphaForPacketLossFractionSmoother) {}
+
+  // Gets the smoothed packet loss fraction.
+  float GetAverage() const {
+    float value = smoother_.filtered();
+    return (value == rtc::ExpFilter::kValueUndefined) ? 0.0f : value;
+  }
+
+  // Add new observation to the packet loss fraction smoother.
+  void AddSample(float packet_loss_fraction) {
+    int64_t now_ms = rtc::TimeMillis();
+    smoother_.Apply(static_cast<float>(now_ms - last_sample_time_ms_),
+                    packet_loss_fraction);
+    last_sample_time_ms_ = now_ms;
+  }
+
+ private:
+  int64_t last_sample_time_ms_;
+
+  // An exponential filter is used to smooth the packet loss fraction.
+  rtc::ExpFilter smoother_;
+};
+
+AudioEncoderOpusImpl::AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config,
+                                           int payload_type)
+    : AudioEncoderOpusImpl(
+          config,
+          payload_type,
+          [this](absl::string_view config_string, RtcEventLog* event_log) {
+            return DefaultAudioNetworkAdaptorCreator(config_string, event_log);
+          },
+          // We choose 5sec as initial time constant due to empirical data.
+          std::make_unique<SmoothingFilterImpl>(5000)) {}
+
+AudioEncoderOpusImpl::AudioEncoderOpusImpl(
+    const AudioEncoderOpusConfig& config,
+    int payload_type,
+    const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
+    std::unique_ptr<SmoothingFilter> bitrate_smoother)
+    : payload_type_(payload_type),
+      use_stable_target_for_adaptation_(!webrtc::field_trial::IsDisabled(
+          "WebRTC-Audio-StableTargetAdaptation")),
+      adjust_bandwidth_(
+          webrtc::field_trial::IsEnabled("WebRTC-AdjustOpusBandwidth")),
+      bitrate_changed_(true),
+      bitrate_multipliers_(GetBitrateMultipliers()),
+      packet_loss_rate_(0.0),
+      inst_(nullptr),
+      packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
+      audio_network_adaptor_creator_(audio_network_adaptor_creator),
+      bitrate_smoother_(std::move(bitrate_smoother)),
+      consecutive_dtx_frames_(0) {
+  RTC_DCHECK(0 <= payload_type && payload_type <= 127);
+
+  // Sanity check of the redundant payload type field that we want to get rid
+  // of. See https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+  RTC_CHECK(config.payload_type == -1 || config.payload_type == payload_type);
+
+  RTC_CHECK(RecreateEncoderInstance(config));
+  SetProjectedPacketLossRate(packet_loss_rate_);
+}
+
+AudioEncoderOpusImpl::AudioEncoderOpusImpl(int payload_type,
+                                           const SdpAudioFormat& format)
+    : AudioEncoderOpusImpl(*SdpToConfig(format), payload_type) {}
+
+AudioEncoderOpusImpl::~AudioEncoderOpusImpl() {
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+}
+
+int AudioEncoderOpusImpl::SampleRateHz() const {
+  return config_.sample_rate_hz;
+}
+
+size_t AudioEncoderOpusImpl::NumChannels() const {
+  return config_.num_channels;
+}
+
+int AudioEncoderOpusImpl::RtpTimestampRateHz() const {
+  return kRtpTimestampRateHz;
+}
+
+size_t AudioEncoderOpusImpl::Num10MsFramesInNextPacket() const {
+  return Num10msFramesPerPacket();
+}
+
+size_t AudioEncoderOpusImpl::Max10MsFramesInAPacket() const {
+  return Num10msFramesPerPacket();
+}
+
+int AudioEncoderOpusImpl::GetTargetBitrate() const {
+  return GetBitrateBps(config_);
+}
+
+void AudioEncoderOpusImpl::Reset() {
+  RTC_CHECK(RecreateEncoderInstance(config_));
+}
+
+bool AudioEncoderOpusImpl::SetFec(bool enable) {
+  if (enable) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+  }
+  config_.fec_enabled = enable;
+  return true;
+}
+
+bool AudioEncoderOpusImpl::SetDtx(bool enable) {
+  if (enable) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+  }
+  config_.dtx_enabled = enable;
+  return true;
+}
+
+bool AudioEncoderOpusImpl::GetDtx() const {
+  return config_.dtx_enabled;
+}
+
+bool AudioEncoderOpusImpl::SetApplication(Application application) {
+  auto conf = config_;
+  switch (application) {
+    case Application::kSpeech:
+      conf.application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
+      break;
+    case Application::kAudio:
+      conf.application = AudioEncoderOpusConfig::ApplicationMode::kAudio;
+      break;
+  }
+  return RecreateEncoderInstance(conf);
+}
+
+void AudioEncoderOpusImpl::SetMaxPlaybackRate(int frequency_hz) {
+  auto conf = config_;
+  conf.max_playback_rate_hz = frequency_hz;
+  RTC_CHECK(RecreateEncoderInstance(conf));
+}
+
+bool AudioEncoderOpusImpl::EnableAudioNetworkAdaptor(
+    const std::string& config_string,
+    RtcEventLog* event_log) {
+  audio_network_adaptor_ =
+      audio_network_adaptor_creator_(config_string, event_log);
+  return audio_network_adaptor_.get() != nullptr;
+}
+
+void AudioEncoderOpusImpl::DisableAudioNetworkAdaptor() {
+  audio_network_adaptor_.reset(nullptr);
+}
+
+void AudioEncoderOpusImpl::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  if (audio_network_adaptor_) {
+    audio_network_adaptor_->SetUplinkPacketLossFraction(
+        uplink_packet_loss_fraction);
+    ApplyAudioNetworkAdaptor();
+  }
+  packet_loss_fraction_smoother_->AddSample(uplink_packet_loss_fraction);
+  float average_fraction_loss = packet_loss_fraction_smoother_->GetAverage();
+  SetProjectedPacketLossRate(average_fraction_loss);
+}
+
+void AudioEncoderOpusImpl::OnReceivedTargetAudioBitrate(
+    int target_audio_bitrate_bps) {
+  SetTargetBitrate(target_audio_bitrate_bps);
+}
+
+void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms,
+    absl::optional<int64_t> stable_target_bitrate_bps) {
+  if (audio_network_adaptor_) {
+    audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
+    if (use_stable_target_for_adaptation_) {
+      if (stable_target_bitrate_bps)
+        audio_network_adaptor_->SetUplinkBandwidth(*stable_target_bitrate_bps);
+    } else {
+      // We give smoothed bitrate allocation to audio network adaptor as
+      // the uplink bandwidth.
+      // The BWE spikes should not affect the bitrate smoother more than 25%.
+      // To simplify the calculations we use a step response as input signal.
+      // The step response of an exponential filter is
+      // u(t) = 1 - e^(-t / time_constant).
+      // In order to limit the affect of a BWE spike within 25% of its value
+      // before
+      // the next BWE update, we would choose a time constant that fulfills
+      // 1 - e^(-bwe_period_ms / time_constant) < 0.25
+      // Then 4 * bwe_period_ms is a good choice.
+      if (bwe_period_ms)
+        bitrate_smoother_->SetTimeConstantMs(*bwe_period_ms * 4);
+      bitrate_smoother_->AddSample(target_audio_bitrate_bps);
+    }
+
+    ApplyAudioNetworkAdaptor();
+  } else {
+    if (!overhead_bytes_per_packet_) {
+      RTC_LOG(LS_INFO)
+          << "AudioEncoderOpusImpl: Overhead unknown, target audio bitrate "
+          << target_audio_bitrate_bps << " bps is ignored.";
+      return;
+    }
+    const int overhead_bps = static_cast<int>(
+        *overhead_bytes_per_packet_ * 8 * 100 / Num10MsFramesInNextPacket());
+    SetTargetBitrate(
+        std::min(AudioEncoderOpusConfig::kMaxBitrateBps,
+                 std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                          target_audio_bitrate_bps - overhead_bps)));
+  }
+}
+void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {
+  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, bwe_period_ms,
+                            absl::nullopt);
+}
+
+void AudioEncoderOpusImpl::OnReceivedUplinkAllocation(
+    BitrateAllocationUpdate update) {
+  OnReceivedUplinkBandwidth(update.target_bitrate.bps(), update.bwe_period.ms(),
+                            update.stable_target_bitrate.bps());
+}
+
+void AudioEncoderOpusImpl::OnReceivedRtt(int rtt_ms) {
+  if (!audio_network_adaptor_)
+    return;
+  audio_network_adaptor_->SetRtt(rtt_ms);
+  ApplyAudioNetworkAdaptor();
+}
+
+void AudioEncoderOpusImpl::OnReceivedOverhead(
+    size_t overhead_bytes_per_packet) {
+  if (audio_network_adaptor_) {
+    audio_network_adaptor_->SetOverhead(overhead_bytes_per_packet);
+    ApplyAudioNetworkAdaptor();
+  } else {
+    overhead_bytes_per_packet_ = overhead_bytes_per_packet;
+  }
+}
+
+void AudioEncoderOpusImpl::SetReceiverFrameLengthRange(
+    int min_frame_length_ms,
+    int max_frame_length_ms) {
+  // Ensure that `SetReceiverFrameLengthRange` is called before
+  // `EnableAudioNetworkAdaptor`, otherwise we need to recreate
+  // `audio_network_adaptor_`, which is not a needed use case.
+  RTC_DCHECK(!audio_network_adaptor_);
+  FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
+                            &config_.supported_frame_lengths_ms);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  MaybeUpdateUplinkBandwidth();
+
+  if (input_buffer_.empty())
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
+  if (input_buffer_.size() <
+      (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(input_buffer_.size(),
+               Num10msFramesPerPacket() * SamplesPer10msFrame());
+
+  const size_t max_encoded_bytes = SufficientOutputBufferSize();
+  EncodedInfo info;
+  info.encoded_bytes = encoded->AppendData(
+      max_encoded_bytes, [&](rtc::ArrayView<uint8_t> encoded) {
+        int status = WebRtcOpus_Encode(
+            inst_, &input_buffer_[0],
+            rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
+            rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded.data());
+
+        RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
+
+        return static_cast<size_t>(status);
+      });
+  input_buffer_.clear();
+
+  bool dtx_frame = (info.encoded_bytes <= 2);
+
+  // Will use new packet size for next encoding.
+  config_.frame_size_ms = next_frame_length_ms_;
+
+  if (adjust_bandwidth_ && bitrate_changed_) {
+    const auto bandwidth = GetNewBandwidth(config_, inst_);
+    if (bandwidth) {
+      RTC_CHECK_EQ(0, WebRtcOpus_SetBandwidth(inst_, *bandwidth));
+    }
+    bitrate_changed_ = false;
+  }
+
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.send_even_if_empty = true;  // Allows Opus to send empty packets.
+  // After 20 DTX frames (MAX_CONSECUTIVE_DTX) Opus will send a frame
+  // coding the background noise. Avoid flagging this frame as speech
+  // (even though there is a probability of the frame being speech).
+  info.speech = !dtx_frame && (consecutive_dtx_frames_ != 20);
+  info.encoder_type = CodecType::kOpus;
+
+  // Increase or reset DTX counter.
+  consecutive_dtx_frames_ = (dtx_frame) ? (consecutive_dtx_frames_ + 1) : (0);
+
+  return info;
+}
+
+size_t AudioEncoderOpusImpl::Num10msFramesPerPacket() const {
+  return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
+}
+
+size_t AudioEncoderOpusImpl::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(config_.sample_rate_hz, 100) *
+         config_.num_channels;
+}
+
+size_t AudioEncoderOpusImpl::SufficientOutputBufferSize() const {
+  // Calculate the number of bytes we expect the encoder to produce,
+  // then multiply by two to give a wide margin for error.
+  const size_t bytes_per_millisecond =
+      static_cast<size_t>(GetBitrateBps(config_) / (1000 * 8) + 1);
+  const size_t approx_encoded_bytes =
+      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
+  return 2 * approx_encoded_bytes;
+}
+
+// If the given config is OK, recreate the Opus encoder instance with those
+// settings, save the config, and return true. Otherwise, do nothing and return
+// false.
+bool AudioEncoderOpusImpl::RecreateEncoderInstance(
+    const AudioEncoderOpusConfig& config) {
+  if (!config.IsOk())
+    return false;
+  config_ = config;
+  if (inst_)
+    RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+  input_buffer_.clear();
+  input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(
+                      &inst_, config.num_channels,
+                      config.application ==
+                              AudioEncoderOpusConfig::ApplicationMode::kVoip
+                          ? 0
+                          : 1,
+                      config.sample_rate_hz));
+  const int bitrate = GetBitrateBps(config);
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, bitrate));
+  RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
+  if (config.fec_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+  }
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
+  // Use the default complexity if the start bitrate is within the hysteresis
+  // window.
+  complexity_ = GetNewComplexity(config).value_or(config.complexity);
+  RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
+  bitrate_changed_ = true;
+  if (config.dtx_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+  }
+  RTC_CHECK_EQ(0,
+               WebRtcOpus_SetPacketLossRate(
+                   inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  if (config.cbr_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableCbr(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableCbr(inst_));
+  }
+  num_channels_to_encode_ = NumChannels();
+  next_frame_length_ms_ = config_.frame_size_ms;
+  return true;
+}
+
+void AudioEncoderOpusImpl::SetFrameLength(int frame_length_ms) {
+  if (next_frame_length_ms_ != frame_length_ms) {
+    RTC_LOG(LS_VERBOSE) << "Update Opus frame length "
+                        << "from " << next_frame_length_ms_ << " ms "
+                        << "to " << frame_length_ms << " ms.";
+  }
+  next_frame_length_ms_ = frame_length_ms;
+}
+
+void AudioEncoderOpusImpl::SetNumChannelsToEncode(
+    size_t num_channels_to_encode) {
+  RTC_DCHECK_GT(num_channels_to_encode, 0);
+  RTC_DCHECK_LE(num_channels_to_encode, config_.num_channels);
+
+  if (num_channels_to_encode_ == num_channels_to_encode)
+    return;
+
+  RTC_CHECK_EQ(0, WebRtcOpus_SetForceChannels(inst_, num_channels_to_encode));
+  num_channels_to_encode_ = num_channels_to_encode;
+}
+
+void AudioEncoderOpusImpl::SetProjectedPacketLossRate(float fraction) {
+  fraction = std::min(std::max(fraction, 0.0f), kMaxPacketLossFraction);
+  if (packet_loss_rate_ != fraction) {
+    packet_loss_rate_ = fraction;
+    RTC_CHECK_EQ(
+        0, WebRtcOpus_SetPacketLossRate(
+               inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  }
+}
+
+void AudioEncoderOpusImpl::SetTargetBitrate(int bits_per_second) {
+  const int new_bitrate = rtc::SafeClamp<int>(
+      bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
+      AudioEncoderOpusConfig::kMaxBitrateBps);
+  if (config_.bitrate_bps && *config_.bitrate_bps != new_bitrate) {
+    config_.bitrate_bps = new_bitrate;
+    RTC_DCHECK(config_.IsOk());
+    const int bitrate = GetBitrateBps(config_);
+    RTC_CHECK_EQ(
+        0, WebRtcOpus_SetBitRate(
+               inst_, GetMultipliedBitrate(bitrate, bitrate_multipliers_)));
+    RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
+    bitrate_changed_ = true;
+  }
+
+  const auto new_complexity = GetNewComplexity(config_);
+  if (new_complexity && complexity_ != *new_complexity) {
+    complexity_ = *new_complexity;
+    RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
+  }
+}
+
+void AudioEncoderOpusImpl::ApplyAudioNetworkAdaptor() {
+  auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
+
+  if (config.bitrate_bps)
+    SetTargetBitrate(*config.bitrate_bps);
+  if (config.frame_length_ms)
+    SetFrameLength(*config.frame_length_ms);
+  if (config.enable_dtx)
+    SetDtx(*config.enable_dtx);
+  if (config.num_channels)
+    SetNumChannelsToEncode(*config.num_channels);
+}
+
+std::unique_ptr<AudioNetworkAdaptor>
+AudioEncoderOpusImpl::DefaultAudioNetworkAdaptorCreator(
+    absl::string_view config_string,
+    RtcEventLog* event_log) const {
+  AudioNetworkAdaptorImpl::Config config;
+  config.event_log = event_log;
+  return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
+      config, ControllerManagerImpl::Create(
+                  config_string, NumChannels(), supported_frame_lengths_ms(),
+                  AudioEncoderOpusConfig::kMinBitrateBps,
+                  num_channels_to_encode_, next_frame_length_ms_,
+                  GetTargetBitrate(), config_.fec_enabled, GetDtx())));
+}
+
+void AudioEncoderOpusImpl::MaybeUpdateUplinkBandwidth() {
+  if (audio_network_adaptor_ && !use_stable_target_for_adaptation_) {
+    int64_t now_ms = rtc::TimeMillis();
+    if (!bitrate_smoother_last_update_time_ ||
+        now_ms - *bitrate_smoother_last_update_time_ >=
+            config_.uplink_bandwidth_update_interval_ms) {
+      absl::optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
+      if (smoothed_bitrate)
+        audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
+      bitrate_smoother_last_update_time_ = now_ms;
+    }
+  }
+}
+
+ANAStats AudioEncoderOpusImpl::GetANAStats() const {
+  if (audio_network_adaptor_) {
+    return audio_network_adaptor_->GetStats();
+  }
+  return ANAStats();
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta> >
+AudioEncoderOpusImpl::GetFrameLengthRange() const {
+  if (audio_network_adaptor_) {
+    if (config_.supported_frame_lengths_ms.empty()) {
+      return absl::nullopt;
+    }
+    return {{TimeDelta::Millis(config_.supported_frame_lengths_ms.front()),
+             TimeDelta::Millis(config_.supported_frame_lengths_ms.back())}};
+  } else {
+    return {{TimeDelta::Millis(config_.frame_size_ms),
+             TimeDelta::Millis(config_.frame_size_ms)}};
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
new file mode 100644
index 0000000000..8c5c235016
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+
+#include <functional>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "common_audio/smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class AudioEncoderOpusImpl final : public AudioEncoder {
+ public:
+  // Returns empty if the current bitrate falls within the hysteresis window,
+  // defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
+  // Otherwise, returns the current complexity depending on whether the
+  // current bitrate is above or below complexity_threshold_bps.
+  static absl::optional<int> GetNewComplexity(
+      const AudioEncoderOpusConfig& config);
+
+  // Returns OPUS_AUTO if the the current bitrate is above wideband threshold.
+  // Returns empty if it is below, but bandwidth coincides with the desired one.
+  // Otherwise returns the desired bandwidth.
+  static absl::optional<int> GetNewBandwidth(
+      const AudioEncoderOpusConfig& config,
+      OpusEncInst* inst);
+
+  using AudioNetworkAdaptorCreator =
+      std::function<std::unique_ptr<AudioNetworkAdaptor>(absl::string_view,
+                                                         RtcEventLog*)>;
+
+  AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config, int payload_type);
+
+  // Dependency injection for testing.
+  AudioEncoderOpusImpl(
+      const AudioEncoderOpusConfig& config,
+      int payload_type,
+      const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
+      std::unique_ptr<SmoothingFilter> bitrate_smoother);
+
+  AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format);
+  ~AudioEncoderOpusImpl() override;
+
+  AudioEncoderOpusImpl(const AudioEncoderOpusImpl&) = delete;
+  AudioEncoderOpusImpl& operator=(const AudioEncoderOpusImpl&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  void Reset() override;
+  bool SetFec(bool enable) override;
+
+  // Set Opus DTX. Once enabled, Opus stops transmission, when it detects
+  // voice being inactive. During that, it still sends 2 packets (one for
+  // content, one for signaling) about every 400 ms.
+  bool SetDtx(bool enable) override;
+  bool GetDtx() const override;
+
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  bool EnableAudioNetworkAdaptor(const std::string& config_string,
+                                 RtcEventLog* event_log) override;
+  void DisableAudioNetworkAdaptor() override;
+  void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) override;
+  void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms) override;
+  void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
+  void OnReceivedRtt(int rtt_ms) override;
+  void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
+  void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                   int max_frame_length_ms) override;
+  ANAStats GetANAStats() const override;
+  absl::optional<std::pair<TimeDelta, TimeDelta> > GetFrameLengthRange()
+      const override;
+  rtc::ArrayView<const int> supported_frame_lengths_ms() const {
+    return config_.supported_frame_lengths_ms;
+  }
+
+  // Getters for testing.
+  float packet_loss_rate() const { return packet_loss_rate_; }
+  AudioEncoderOpusConfig::ApplicationMode application() const {
+    return config_.application;
+  }
+  bool fec_enabled() const { return config_.fec_enabled; }
+  size_t num_channels_to_encode() const { return num_channels_to_encode_; }
+  int next_frame_length_ms() const { return next_frame_length_ms_; }
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  class PacketLossFractionSmoother;
+
+  static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderOpusConfig&,
+      int payload_type);
+
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
+  size_t SufficientOutputBufferSize() const;
+  bool RecreateEncoderInstance(const AudioEncoderOpusConfig& config);
+  void SetFrameLength(int frame_length_ms);
+  void SetNumChannelsToEncode(size_t num_channels_to_encode);
+  void SetProjectedPacketLossRate(float fraction);
+
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms,
+      absl::optional<int64_t> link_capacity_allocation);
+
+  // TODO(minyue): remove "override" when we can deprecate
+  // `AudioEncoder::SetTargetBitrate`.
+  void SetTargetBitrate(int target_bps) override;
+
+  void ApplyAudioNetworkAdaptor();
+  std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
+      absl::string_view config_string,
+      RtcEventLog* event_log) const;
+
+  void MaybeUpdateUplinkBandwidth();
+
+  AudioEncoderOpusConfig config_;
+  const int payload_type_;
+  const bool use_stable_target_for_adaptation_;
+  const bool adjust_bandwidth_;
+  bool bitrate_changed_;
+  // A multiplier for bitrates at 5 kbps and higher. The target bitrate
+  // will be multiplied by these multipliers, each multiplier is applied to a
+  // 1 kbps range.
+  std::vector<float> bitrate_multipliers_;
+  float packet_loss_rate_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+  size_t num_channels_to_encode_;
+  int next_frame_length_ms_;
+  int complexity_;
+  std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
+  const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
+  std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
+  absl::optional<size_t> overhead_bytes_per_packet_;
+  const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
+  absl::optional<int64_t> bitrate_smoother_last_update_time_;
+  int consecutive_dtx_frames_;
+
+  friend struct AudioEncoderOpus;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
new file mode 100644
index 0000000000..a2ebe43bbe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -0,0 +1,914 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+
+#include <array>
+#include <memory>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/mocks/mock_smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/fake_clock.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace {
+
+constexpr int kDefaultOpusPayloadType = 105;
+constexpr int kDefaultOpusRate = 32000;
+constexpr int kDefaultOpusPacSize = 960;
+constexpr int64_t kInitialTimeUs = 12345678;
+
+AudioEncoderOpusConfig CreateConfigWithParameters(
+    const SdpAudioFormat::Parameters& params) {
+  const SdpAudioFormat format("opus", 48000, 2, params);
+  return *AudioEncoderOpus::SdpToConfig(format);
+}
+
+struct AudioEncoderOpusStates {
+  MockAudioNetworkAdaptor* mock_audio_network_adaptor;
+  MockSmoothingFilter* mock_bitrate_smoother;
+  std::unique_ptr<AudioEncoderOpusImpl> encoder;
+  std::unique_ptr<rtc::ScopedFakeClock> fake_clock;
+  AudioEncoderOpusConfig config;
+};
+
+std::unique_ptr<AudioEncoderOpusStates> CreateCodec(int sample_rate_hz,
+                                                    size_t num_channels) {
+  std::unique_ptr<AudioEncoderOpusStates> states =
+      std::make_unique<AudioEncoderOpusStates>();
+  states->mock_audio_network_adaptor = nullptr;
+  states->fake_clock.reset(new rtc::ScopedFakeClock());
+  states->fake_clock->SetTime(Timestamp::Micros(kInitialTimeUs));
+
+  MockAudioNetworkAdaptor** mock_ptr = &states->mock_audio_network_adaptor;
+  AudioEncoderOpusImpl::AudioNetworkAdaptorCreator creator =
+      [mock_ptr](absl::string_view, RtcEventLog* event_log) {
+        std::unique_ptr<MockAudioNetworkAdaptor> adaptor(
+            new NiceMock<MockAudioNetworkAdaptor>());
+        EXPECT_CALL(*adaptor, Die());
+        *mock_ptr = adaptor.get();
+        return adaptor;
+      };
+
+  AudioEncoderOpusConfig config;
+  config.frame_size_ms = rtc::CheckedDivExact(kDefaultOpusPacSize, 48);
+  config.sample_rate_hz = sample_rate_hz;
+  config.num_channels = num_channels;
+  config.bitrate_bps = kDefaultOpusRate;
+  config.application = num_channels == 1
+                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
+  config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
+  states->config = config;
+
+  std::unique_ptr<MockSmoothingFilter> bitrate_smoother(
+      new MockSmoothingFilter());
+  states->mock_bitrate_smoother = bitrate_smoother.get();
+
+  states->encoder.reset(
+      new AudioEncoderOpusImpl(states->config, kDefaultOpusPayloadType, creator,
+                               std::move(bitrate_smoother)));
+  return states;
+}
+
+AudioEncoderRuntimeConfig CreateEncoderRuntimeConfig() {
+  constexpr int kBitrate = 40000;
+  constexpr int kFrameLength = 60;
+  constexpr bool kEnableDtx = false;
+  constexpr size_t kNumChannels = 1;
+  AudioEncoderRuntimeConfig config;
+  config.bitrate_bps = kBitrate;
+  config.frame_length_ms = kFrameLength;
+  config.enable_dtx = kEnableDtx;
+  config.num_channels = kNumChannels;
+  return config;
+}
+
+void CheckEncoderRuntimeConfig(const AudioEncoderOpusImpl* encoder,
+                               const AudioEncoderRuntimeConfig& config) {
+  EXPECT_EQ(*config.bitrate_bps, encoder->GetTargetBitrate());
+  EXPECT_EQ(*config.frame_length_ms, encoder->next_frame_length_ms());
+  EXPECT_EQ(*config.enable_dtx, encoder->GetDtx());
+  EXPECT_EQ(*config.num_channels, encoder->num_channels_to_encode());
+}
+
+// Create 10ms audio data blocks for a total packet size of "packet_size_ms".
+std::unique_ptr<test::AudioLoop> Create10msAudioBlocks(
+    const std::unique_ptr<AudioEncoderOpusImpl>& encoder,
+    int packet_size_ms) {
+  const std::string file_name =
+      test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+
+  std::unique_ptr<test::AudioLoop> speech_data(new test::AudioLoop());
+  int audio_samples_per_ms =
+      rtc::CheckedDivExact(encoder->SampleRateHz(), 1000);
+  if (!speech_data->Init(
+          file_name,
+          packet_size_ms * audio_samples_per_ms *
+              encoder->num_channels_to_encode(),
+          10 * audio_samples_per_ms * encoder->num_channels_to_encode()))
+    return nullptr;
+  return speech_data;
+}
+
+}  // namespace
+
+class AudioEncoderOpusTest : public ::testing::TestWithParam<int> {
+ protected:
+  int sample_rate_hz_{GetParam()};
+};
+INSTANTIATE_TEST_SUITE_P(Param,
+                         AudioEncoderOpusTest,
+                         ::testing::Values(16000, 48000));
+
+TEST_P(AudioEncoderOpusTest, DefaultApplicationModeMono) {
+  auto states = CreateCodec(sample_rate_hz_, 1);
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, ChangeApplicationMode) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  EXPECT_TRUE(
+      states->encoder->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  // Trigger a reset.
+  states->encoder->Reset();
+  // Verify that the mode is still kAudio.
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
+            states->encoder->application());
+
+  // Now change to kVoip.
+  EXPECT_TRUE(
+      states->encoder->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+
+  // Trigger a reset again.
+  states->encoder->Reset();
+  // Verify that the mode is still kVoip.
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, ToggleDtx) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  // Enable DTX
+  EXPECT_TRUE(states->encoder->SetDtx(true));
+  EXPECT_TRUE(states->encoder->GetDtx());
+  // Turn off DTX.
+  EXPECT_TRUE(states->encoder->SetDtx(false));
+  EXPECT_FALSE(states->encoder->GetDtx());
+}
+
+TEST_P(AudioEncoderOpusTest,
+       OnReceivedUplinkBandwidthWithoutAudioNetworkAdaptor) {
+  auto states = CreateCodec(sample_rate_hz_, 1);
+  // Constants are replicated from audio_states->encoderopus.cc.
+  const int kMinBitrateBps = 6000;
+  const int kMaxBitrateBps = 510000;
+  const int kOverheadBytesPerPacket = 64;
+  states->encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
+  const int kOverheadBps = 8 * kOverheadBytesPerPacket *
+                           rtc::CheckedDivExact(48000, kDefaultOpusPacSize);
+  // Set a too low bitrate.
+  states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps + kOverheadBps - 1,
+                                             absl::nullopt);
+  EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
+  // Set a too high bitrate.
+  states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + kOverheadBps + 1,
+                                             absl::nullopt);
+  EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
+  // Set the minimum rate.
+  states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps + kOverheadBps,
+                                             absl::nullopt);
+  EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
+  // Set the maximum rate.
+  states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + kOverheadBps,
+                                             absl::nullopt);
+  EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
+  // Set rates from kMaxBitrateBps up to 32000 bps.
+  for (int rate = kMinBitrateBps + kOverheadBps; rate <= 32000 + kOverheadBps;
+       rate += 1000) {
+    states->encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
+    EXPECT_EQ(rate - kOverheadBps, states->encoder->GetTargetBitrate());
+  }
+}
+
+TEST_P(AudioEncoderOpusTest, SetReceiverFrameLengthRange) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  // Before calling to `SetReceiverFrameLengthRange`,
+  // `supported_frame_lengths_ms` should contain only the frame length being
+  // used.
+  using ::testing::ElementsAre;
+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+              ElementsAre(states->encoder->next_frame_length_ms()));
+  states->encoder->SetReceiverFrameLengthRange(0, 12345);
+  states->encoder->SetReceiverFrameLengthRange(21, 60);
+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+              ElementsAre(40, 60));
+  states->encoder->SetReceiverFrameLengthRange(20, 59);
+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+              ElementsAre(20, 40));
+}
+
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkPacketLossFraction) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any packet loss fraction is fine.
+  constexpr float kUplinkPacketLoss = 0.1f;
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetUplinkPacketLossFraction(kUplinkPacketLoss));
+  states->encoder->OnReceivedUplinkPacketLossFraction(kUplinkPacketLoss);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-StableTargetAdaptation/Disabled/");
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any target audio bitrate is fine.
+  constexpr int kTargetAudioBitrate = 30000;
+  constexpr int64_t kProbingIntervalMs = 3000;
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetTargetAudioBitrate(kTargetAudioBitrate));
+  EXPECT_CALL(*states->mock_bitrate_smoother,
+              SetTimeConstantMs(kProbingIntervalMs * 4));
+  EXPECT_CALL(*states->mock_bitrate_smoother, AddSample(kTargetAudioBitrate));
+  states->encoder->OnReceivedUplinkBandwidth(kTargetAudioBitrate,
+                                             kProbingIntervalMs);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkAllocation) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  BitrateAllocationUpdate update;
+  update.target_bitrate = DataRate::BitsPerSec(30000);
+  update.stable_target_bitrate = DataRate::BitsPerSec(20000);
+  update.bwe_period = TimeDelta::Millis(200);
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetTargetAudioBitrate(update.target_bitrate.bps()));
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetUplinkBandwidth(update.stable_target_bitrate.bps()));
+  states->encoder->OnReceivedUplinkAllocation(update);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedRtt) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any rtt is fine.
+  constexpr int kRtt = 30;
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetRtt(kRtt));
+  states->encoder->OnReceivedRtt(kRtt);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedOverhead) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any overhead is fine.
+  constexpr size_t kOverhead = 64;
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetOverhead(kOverhead));
+  states->encoder->OnReceivedOverhead(kOverhead);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest,
+       PacketLossFractionSmoothedOnSetUplinkPacketLossFraction) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  // The values are carefully chosen so that if no smoothing is made, the test
+  // will fail.
+  constexpr float kPacketLossFraction_1 = 0.02f;
+  constexpr float kPacketLossFraction_2 = 0.198f;
+  // `kSecondSampleTimeMs` is chosen to ease the calculation since
+  // 0.9999 ^ 6931 = 0.5.
+  constexpr int64_t kSecondSampleTimeMs = 6931;
+
+  // First time, no filtering.
+  states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_1);
+  EXPECT_FLOAT_EQ(0.02f, states->encoder->packet_loss_rate());
+
+  states->fake_clock->AdvanceTime(TimeDelta::Millis(kSecondSampleTimeMs));
+  states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_2);
+
+  // Now the output of packet loss fraction smoother should be
+  // (0.02 + 0.198) / 2 = 0.109.
+  EXPECT_NEAR(0.109f, states->encoder->packet_loss_rate(), 0.001);
+}
+
+TEST_P(AudioEncoderOpusTest, PacketLossRateUpperBounded) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  states->encoder->OnReceivedUplinkPacketLossFraction(0.5);
+  EXPECT_FLOAT_EQ(0.2f, states->encoder->packet_loss_rate());
+}
+
+TEST_P(AudioEncoderOpusTest, DoNotInvokeSetTargetBitrateIfOverheadUnknown) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusRate * 2,
+                                             absl::nullopt);
+
+  // Since `OnReceivedOverhead` has not been called, the codec bitrate should
+  // not change.
+  EXPECT_EQ(kDefaultOpusRate, states->encoder->GetTargetBitrate());
+}
+
+// Verifies that the complexity adaptation in the config works as intended.
+TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
+  AudioEncoderOpusConfig config;
+  config.low_rate_complexity = 8;
+  config.complexity = 6;
+
+  // Bitrate within hysteresis window. Expect empty output.
+  config.bitrate_bps = 12500;
+  EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
+
+  // Bitrate below hysteresis window. Expect higher complexity.
+  config.bitrate_bps = 10999;
+  EXPECT_EQ(8, AudioEncoderOpusImpl::GetNewComplexity(config));
+
+  // Bitrate within hysteresis window. Expect empty output.
+  config.bitrate_bps = 12500;
+  EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
+
+  // Bitrate above hysteresis window. Expect lower complexity.
+  config.bitrate_bps = 14001;
+  EXPECT_EQ(6, AudioEncoderOpusImpl::GetNewComplexity(config));
+}
+
+// Verifies that the bandwidth adaptation in the config works as intended.
+TEST_P(AudioEncoderOpusTest, ConfigBandwidthAdaptation) {
+  AudioEncoderOpusConfig config;
+  const size_t opus_rate_khz = rtc::CheckedDivExact(sample_rate_hz_, 1000);
+  const std::vector<int16_t> silence(
+      opus_rate_khz * config.frame_size_ms * config.num_channels, 0);
+  constexpr size_t kMaxBytes = 1000;
+  uint8_t bitstream[kMaxBytes];
+
+  OpusEncInst* inst;
+  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(
+                   &inst, config.num_channels,
+                   config.application ==
+                           AudioEncoderOpusConfig::ApplicationMode::kVoip
+                       ? 0
+                       : 1,
+                   sample_rate_hz_));
+
+  // Bitrate below minmum wideband. Expect narrowband.
+  config.bitrate_bps = absl::optional<int>(7999);
+  auto bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND), bandwidth);
+  WebRtcOpus_SetBandwidth(inst, *bandwidth);
+  // It is necessary to encode here because Opus has some logic in the encoder
+  // that goes from the user-set bandwidth to the used and returned one.
+  WebRtcOpus_Encode(inst, silence.data(),
+                    rtc::CheckedDivExact(silence.size(), config.num_channels),
+                    kMaxBytes, bitstream);
+
+  // Bitrate not yet above maximum narrowband. Expect empty.
+  config.bitrate_bps = absl::optional<int>(9000);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(), bandwidth);
+
+  // Bitrate above maximum narrowband. Expect wideband.
+  config.bitrate_bps = absl::optional<int>(9001);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND), bandwidth);
+  WebRtcOpus_SetBandwidth(inst, *bandwidth);
+  // It is necessary to encode here because Opus has some logic in the encoder
+  // that goes from the user-set bandwidth to the used and returned one.
+  WebRtcOpus_Encode(inst, silence.data(),
+                    rtc::CheckedDivExact(silence.size(), config.num_channels),
+                    kMaxBytes, bitstream);
+
+  // Bitrate not yet below minimum wideband. Expect empty.
+  config.bitrate_bps = absl::optional<int>(8000);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(), bandwidth);
+
+  // Bitrate above automatic threshold. Expect automatic.
+  config.bitrate_bps = absl::optional<int>(12001);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(OPUS_AUTO), bandwidth);
+
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(inst));
+}
+
+TEST_P(AudioEncoderOpusTest, EmptyConfigDoesNotAffectEncoderSettings) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig empty_config;
+
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config))
+      .WillOnce(Return(empty_config));
+
+  constexpr size_t kOverhead = 64;
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetOverhead(kOverhead))
+      .Times(2);
+  states->encoder->OnReceivedOverhead(kOverhead);
+  states->encoder->OnReceivedOverhead(kOverhead);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-StableTargetAdaptation/Disabled/");
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+  const size_t opus_rate_khz = rtc::CheckedDivExact(sample_rate_hz_, 1000);
+  const std::vector<int16_t> audio(opus_rate_khz * 10 * 2, 0);
+  rtc::Buffer encoded;
+  EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
+      .WillOnce(Return(50000));
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(50000));
+  states->encoder->Encode(
+      0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+
+  // Repeat update uplink bandwidth tests.
+  for (int i = 0; i < 5; i++) {
+    // Don't update till it is time to update again.
+    states->fake_clock->AdvanceTime(TimeDelta::Millis(
+        states->config.uplink_bandwidth_update_interval_ms - 1));
+    states->encoder->Encode(
+        0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+
+    // Update when it is time to update.
+    EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
+        .WillOnce(Return(40000));
+    EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(40000));
+    states->fake_clock->AdvanceTime(TimeDelta::Millis(1));
+    states->encoder->Encode(
+        0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+  }
+}
+
+TEST_P(AudioEncoderOpusTest, EncodeAtMinBitrate) {
+  auto states = CreateCodec(sample_rate_hz_, 1);
+  constexpr int kNumPacketsToEncode = 2;
+  auto audio_frames =
+      Create10msAudioBlocks(states->encoder, kNumPacketsToEncode * 20);
+  ASSERT_TRUE(audio_frames) << "Create10msAudioBlocks failed";
+  rtc::Buffer encoded;
+  uint32_t rtp_timestamp = 12345;  // Just a number not important to this test.
+
+  states->encoder->OnReceivedUplinkBandwidth(0, absl::nullopt);
+  for (int packet_index = 0; packet_index < kNumPacketsToEncode;
+       packet_index++) {
+    // Make sure we are not encoding before we have enough data for
+    // a 20ms packet.
+    for (int index = 0; index < 1; index++) {
+      states->encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
+                              &encoded);
+      EXPECT_EQ(0u, encoded.size());
+    }
+
+    // Should encode now.
+    states->encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
+                            &encoded);
+    EXPECT_GT(encoded.size(), 0u);
+    encoded.Clear();
+  }
+}
+
+TEST(AudioEncoderOpusTest, TestConfigDefaults) {
+  const auto config_opt = AudioEncoderOpus::SdpToConfig({"opus", 48000, 2});
+  ASSERT_TRUE(config_opt);
+  EXPECT_EQ(48000, config_opt->max_playback_rate_hz);
+  EXPECT_EQ(1u, config_opt->num_channels);
+  EXPECT_FALSE(config_opt->fec_enabled);
+  EXPECT_FALSE(config_opt->dtx_enabled);
+  EXPECT_EQ(20, config_opt->frame_size_ms);
+}
+
+TEST(AudioEncoderOpusTest, TestConfigFromParams) {
+  const auto config1 = CreateConfigWithParameters({{"stereo", "0"}});
+  EXPECT_EQ(1U, config1.num_channels);
+
+  const auto config2 = CreateConfigWithParameters({{"stereo", "1"}});
+  EXPECT_EQ(2U, config2.num_channels);
+
+  const auto config3 = CreateConfigWithParameters({{"useinbandfec", "0"}});
+  EXPECT_FALSE(config3.fec_enabled);
+
+  const auto config4 = CreateConfigWithParameters({{"useinbandfec", "1"}});
+  EXPECT_TRUE(config4.fec_enabled);
+
+  const auto config5 = CreateConfigWithParameters({{"usedtx", "0"}});
+  EXPECT_FALSE(config5.dtx_enabled);
+
+  const auto config6 = CreateConfigWithParameters({{"usedtx", "1"}});
+  EXPECT_TRUE(config6.dtx_enabled);
+
+  const auto config7 = CreateConfigWithParameters({{"cbr", "0"}});
+  EXPECT_FALSE(config7.cbr_enabled);
+
+  const auto config8 = CreateConfigWithParameters({{"cbr", "1"}});
+  EXPECT_TRUE(config8.cbr_enabled);
+
+  const auto config9 =
+      CreateConfigWithParameters({{"maxplaybackrate", "12345"}});
+  EXPECT_EQ(12345, config9.max_playback_rate_hz);
+
+  const auto config10 =
+      CreateConfigWithParameters({{"maxaveragebitrate", "96000"}});
+  EXPECT_EQ(96000, config10.bitrate_bps);
+
+  const auto config11 = CreateConfigWithParameters({{"maxptime", "40"}});
+  for (int frame_length : config11.supported_frame_lengths_ms) {
+    EXPECT_LE(frame_length, 40);
+  }
+
+  const auto config12 = CreateConfigWithParameters({{"minptime", "40"}});
+  for (int frame_length : config12.supported_frame_lengths_ms) {
+    EXPECT_GE(frame_length, 40);
+  }
+
+  const auto config13 = CreateConfigWithParameters({{"ptime", "40"}});
+  EXPECT_EQ(40, config13.frame_size_ms);
+
+  constexpr int kMinSupportedFrameLength = 10;
+  constexpr int kMaxSupportedFrameLength =
+      WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
+
+  const auto config14 = CreateConfigWithParameters({{"ptime", "1"}});
+  EXPECT_EQ(kMinSupportedFrameLength, config14.frame_size_ms);
+
+  const auto config15 = CreateConfigWithParameters({{"ptime", "2000"}});
+  EXPECT_EQ(kMaxSupportedFrameLength, config15.frame_size_ms);
+}
+
+TEST(AudioEncoderOpusTest, TestConfigFromInvalidParams) {
+  const webrtc::SdpAudioFormat format("opus", 48000, 2);
+  const auto default_config = *AudioEncoderOpus::SdpToConfig(format);
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+  const std::vector<int> default_supported_frame_lengths_ms({20, 40, 60, 120});
+#else
+  const std::vector<int> default_supported_frame_lengths_ms({20, 40, 60});
+#endif
+
+  AudioEncoderOpusConfig config;
+  config = CreateConfigWithParameters({{"stereo", "invalid"}});
+  EXPECT_EQ(default_config.num_channels, config.num_channels);
+
+  config = CreateConfigWithParameters({{"useinbandfec", "invalid"}});
+  EXPECT_EQ(default_config.fec_enabled, config.fec_enabled);
+
+  config = CreateConfigWithParameters({{"usedtx", "invalid"}});
+  EXPECT_EQ(default_config.dtx_enabled, config.dtx_enabled);
+
+  config = CreateConfigWithParameters({{"cbr", "invalid"}});
+  EXPECT_EQ(default_config.dtx_enabled, config.dtx_enabled);
+
+  config = CreateConfigWithParameters({{"maxplaybackrate", "0"}});
+  EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
+
+  config = CreateConfigWithParameters({{"maxplaybackrate", "-23"}});
+  EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
+
+  config = CreateConfigWithParameters({{"maxplaybackrate", "not a number!"}});
+  EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "0"}});
+  EXPECT_EQ(6000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "-1000"}});
+  EXPECT_EQ(6000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "1024000"}});
+  EXPECT_EQ(510000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "not a number!"}});
+  EXPECT_EQ(default_config.bitrate_bps, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxptime", "invalid"}});
+  EXPECT_EQ(default_supported_frame_lengths_ms,
+            config.supported_frame_lengths_ms);
+
+  config = CreateConfigWithParameters({{"minptime", "invalid"}});
+  EXPECT_EQ(default_supported_frame_lengths_ms,
+            config.supported_frame_lengths_ms);
+
+  config = CreateConfigWithParameters({{"ptime", "invalid"}});
+  EXPECT_EQ(default_supported_frame_lengths_ms,
+            config.supported_frame_lengths_ms);
+}
+
+TEST(AudioEncoderOpusTest, GetFrameLenghtRange) {
+  AudioEncoderOpusConfig config =
+      CreateConfigWithParameters({{"maxptime", "10"}, {"ptime", "10"}});
+  std::unique_ptr<AudioEncoder> encoder =
+      AudioEncoderOpus::MakeAudioEncoder(config, kDefaultOpusPayloadType);
+  auto ptime = webrtc::TimeDelta::Millis(10);
+  absl::optional<std::pair<webrtc::TimeDelta, webrtc::TimeDelta>> range = {
+      {ptime, ptime}};
+  EXPECT_EQ(encoder->GetFrameLengthRange(), range);
+}
+
+// Test that bitrate will be overridden by the "maxaveragebitrate" parameter.
+// Also test that the "maxaveragebitrate" can't be set to values outside the
+// range of 6000 and 510000
+TEST(AudioEncoderOpusTest, SetSendCodecOpusMaxAverageBitrate) {
+  // Ignore if less than 6000.
+  const auto config1 = AudioEncoderOpus::SdpToConfig(
+      {"opus", 48000, 2, {{"maxaveragebitrate", "5999"}}});
+  EXPECT_EQ(6000, config1->bitrate_bps);
+
+  // Ignore if larger than 510000.
+  const auto config2 = AudioEncoderOpus::SdpToConfig(
+      {"opus", 48000, 2, {{"maxaveragebitrate", "510001"}}});
+  EXPECT_EQ(510000, config2->bitrate_bps);
+
+  const auto config3 = AudioEncoderOpus::SdpToConfig(
+      {"opus", 48000, 2, {{"maxaveragebitrate", "200000"}}});
+  EXPECT_EQ(200000, config3->bitrate_bps);
+}
+
+// Test maxplaybackrate <= 8000 triggers Opus narrow band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateNb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "8000"}});
+  EXPECT_EQ(8000, config.max_playback_rate_hz);
+  EXPECT_EQ(12000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "8000"}, {"stereo", "1"}});
+  EXPECT_EQ(8000, config.max_playback_rate_hz);
+  EXPECT_EQ(24000, config.bitrate_bps);
+}
+
+// Test 8000 < maxplaybackrate <= 12000 triggers Opus medium band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateMb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "8001"}});
+  EXPECT_EQ(8001, config.max_playback_rate_hz);
+  EXPECT_EQ(20000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "8001"}, {"stereo", "1"}});
+  EXPECT_EQ(8001, config.max_playback_rate_hz);
+  EXPECT_EQ(40000, config.bitrate_bps);
+}
+
+// Test 12000 < maxplaybackrate <= 16000 triggers Opus wide band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateWb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "12001"}});
+  EXPECT_EQ(12001, config.max_playback_rate_hz);
+  EXPECT_EQ(20000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "12001"}, {"stereo", "1"}});
+  EXPECT_EQ(12001, config.max_playback_rate_hz);
+  EXPECT_EQ(40000, config.bitrate_bps);
+}
+
+// Test 16000 < maxplaybackrate <= 24000 triggers Opus super wide band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateSwb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "16001"}});
+  EXPECT_EQ(16001, config.max_playback_rate_hz);
+  EXPECT_EQ(32000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "16001"}, {"stereo", "1"}});
+  EXPECT_EQ(16001, config.max_playback_rate_hz);
+  EXPECT_EQ(64000, config.bitrate_bps);
+}
+
+// Test 24000 < maxplaybackrate triggers Opus full band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateFb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "24001"}});
+  EXPECT_EQ(24001, config.max_playback_rate_hz);
+  EXPECT_EQ(32000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "24001"}, {"stereo", "1"}});
+  EXPECT_EQ(24001, config.max_playback_rate_hz);
+  EXPECT_EQ(64000, config.bitrate_bps);
+}
+
+TEST_P(AudioEncoderOpusTest, OpusFlagDtxAsNonSpeech) {
+  // Create encoder with DTX enabled.
+  AudioEncoderOpusConfig config;
+  config.dtx_enabled = true;
+  config.sample_rate_hz = sample_rate_hz_;
+  constexpr int payload_type = 17;
+  const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+
+  // Open file containing speech and silence.
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  test::AudioLoop audio_loop;
+  // Use the file as if it were sampled at our desired input rate.
+  const size_t max_loop_length_samples =
+      sample_rate_hz_ * 10;  // Max 10 second loop.
+  const size_t input_block_size_samples =
+      10 * sample_rate_hz_ / 1000;  // 10 ms.
+  EXPECT_TRUE(audio_loop.Init(kInputFileName, max_loop_length_samples,
+                              input_block_size_samples));
+
+  // Encode.
+  AudioEncoder::EncodedInfo info;
+  rtc::Buffer encoded(500);
+  int nonspeech_frames = 0;
+  int max_nonspeech_frames = 0;
+  int dtx_frames = 0;
+  int max_dtx_frames = 0;
+  uint32_t rtp_timestamp = 0u;
+  for (size_t i = 0; i < 500; ++i) {
+    encoded.Clear();
+
+    // Every second call to the encoder will generate an Opus packet.
+    for (int j = 0; j < 2; j++) {
+      info =
+          encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
+      rtp_timestamp += input_block_size_samples;
+    }
+
+    // Bookkeeping of number of DTX frames.
+    if (info.encoded_bytes <= 2) {
+      ++dtx_frames;
+    } else {
+      if (dtx_frames > max_dtx_frames)
+        max_dtx_frames = dtx_frames;
+      dtx_frames = 0;
+    }
+
+    // Bookkeeping of number of non-speech frames.
+    if (info.speech == 0) {
+      ++nonspeech_frames;
+    } else {
+      if (nonspeech_frames > max_nonspeech_frames)
+        max_nonspeech_frames = nonspeech_frames;
+      nonspeech_frames = 0;
+    }
+  }
+
+  // Maximum number of consecutive non-speech packets should exceed 15.
+  EXPECT_GT(max_nonspeech_frames, 15);
+}
+
+TEST(AudioEncoderOpusTest, OpusDtxFilteringHighEnergyRefreshPackets) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-OpusAvoidNoisePumpingDuringDtx/Enabled/");
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile16kHz", "pcm");
+  constexpr int kSampleRateHz = 16000;
+  AudioEncoderOpusConfig config;
+  config.dtx_enabled = true;
+  config.sample_rate_hz = kSampleRateHz;
+  constexpr int payload_type = 17;
+  const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+  test::AudioLoop audio_loop;
+  constexpr size_t kMaxLoopLengthSaples = kSampleRateHz * 11.6f;
+  constexpr size_t kInputBlockSizeSamples = kSampleRateHz / 100;
+  EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSaples,
+                              kInputBlockSizeSamples));
+  AudioEncoder::EncodedInfo info;
+  rtc::Buffer encoded(500);
+  // Encode the audio file and store the last part that corresponds to silence.
+  constexpr size_t kSilenceDurationSamples = kSampleRateHz * 0.2f;
+  std::array<int16_t, kSilenceDurationSamples> silence;
+  uint32_t rtp_timestamp = 0;
+  bool last_packet_dtx_frame = false;
+  bool opus_entered_dtx = false;
+  bool silence_filled = false;
+  size_t timestamp_start_silence = 0;
+  while (!silence_filled && rtp_timestamp < kMaxLoopLengthSaples) {
+    encoded.Clear();
+    // Every second call to the encoder will generate an Opus packet.
+    for (int j = 0; j < 2; j++) {
+      auto next_frame = audio_loop.GetNextBlock();
+      info = encoder->Encode(rtp_timestamp, next_frame, &encoded);
+      if (opus_entered_dtx) {
+        size_t silence_frame_start = rtp_timestamp - timestamp_start_silence;
+        silence_filled = silence_frame_start >= kSilenceDurationSamples;
+        if (!silence_filled) {
+          std::copy(next_frame.begin(), next_frame.end(),
+                    silence.begin() + silence_frame_start);
+        }
+      }
+      rtp_timestamp += kInputBlockSizeSamples;
+    }
+    EXPECT_TRUE(info.encoded_bytes > 0 || last_packet_dtx_frame);
+    last_packet_dtx_frame = info.encoded_bytes > 0 ? info.encoded_bytes <= 2
+                                                   : last_packet_dtx_frame;
+    if (info.encoded_bytes <= 2 && !opus_entered_dtx) {
+      timestamp_start_silence = rtp_timestamp;
+    }
+    opus_entered_dtx = info.encoded_bytes <= 2;
+  }
+
+  EXPECT_TRUE(silence_filled);
+  // The copied 200 ms of silence is used for creating 6 bursts that are fed to
+  // the encoder, the first three ones with a larger energy and the last three
+  // with a lower energy. This test verifies that the encoder just sends refresh
+  // DTX packets during the last bursts.
+  int number_non_empty_packets_during_increase = 0;
+  int number_non_empty_packets_during_decrease = 0;
+  for (size_t burst = 0; burst < 6; ++burst) {
+    uint32_t rtp_timestamp_start = rtp_timestamp;
+    const bool increase_noise = burst < 3;
+    const float gain = increase_noise ? 1.4f : 0.0f;
+    while (rtp_timestamp < rtp_timestamp_start + kSilenceDurationSamples) {
+      encoded.Clear();
+      // Every second call to the encoder will generate an Opus packet.
+      for (int j = 0; j < 2; j++) {
+        std::array<int16_t, kInputBlockSizeSamples> silence_frame;
+        size_t silence_frame_start = rtp_timestamp - rtp_timestamp_start;
+        std::transform(
+            silence.begin() + silence_frame_start,
+            silence.begin() + silence_frame_start + kInputBlockSizeSamples,
+            silence_frame.begin(), [gain](float s) { return gain * s; });
+        info = encoder->Encode(rtp_timestamp, silence_frame, &encoded);
+        rtp_timestamp += kInputBlockSizeSamples;
+      }
+      EXPECT_TRUE(info.encoded_bytes > 0 || last_packet_dtx_frame);
+      last_packet_dtx_frame = info.encoded_bytes > 0 ? info.encoded_bytes <= 2
+                                                     : last_packet_dtx_frame;
+      // Tracking the number of non empty packets.
+      if (increase_noise && info.encoded_bytes > 2) {
+        number_non_empty_packets_during_increase++;
+      }
+      if (!increase_noise && info.encoded_bytes > 2) {
+        number_non_empty_packets_during_decrease++;
+      }
+    }
+  }
+  // Check that the refresh DTX packets are just sent during the decrease energy
+  // region.
+  EXPECT_EQ(number_non_empty_packets_during_increase, 0);
+  EXPECT_GT(number_non_empty_packets_during_decrease, 0);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
new file mode 100644
index 0000000000..38b60c6187
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "common_audio/include/audio_util.h"
+#include "common_audio/window_generator.h"
+#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kNumChannels = 1u;
+constexpr int kSampleRateHz = 48000;
+constexpr size_t kMaxLoopLengthSamples = kSampleRateHz * 50;  // 50 seconds.
+constexpr size_t kInputBlockSizeSamples = 10 * kSampleRateHz / 1000;   // 10 ms
+constexpr size_t kOutputBlockSizeSamples = 20 * kSampleRateHz / 1000;  // 20 ms
+constexpr size_t kFftSize = 1024;
+constexpr size_t kNarrowbandSize = 4000 * kFftSize / kSampleRateHz;
+constexpr float kKbdAlpha = 1.5f;
+
+class PowerRatioEstimator : public LappedTransform::Callback {
+ public:
+  PowerRatioEstimator() : low_pow_(0.f), high_pow_(0.f) {
+    WindowGenerator::KaiserBesselDerived(kKbdAlpha, kFftSize, window_);
+    transform_.reset(new LappedTransform(kNumChannels, 0u,
+                                         kInputBlockSizeSamples, window_,
+                                         kFftSize, kFftSize / 2, this));
+  }
+
+  void ProcessBlock(float* data) { transform_->ProcessChunk(&data, nullptr); }
+
+  float PowerRatio() { return high_pow_ / low_pow_; }
+
+ protected:
+  void ProcessAudioBlock(const std::complex<float>* const* input,
+                         size_t num_input_channels,
+                         size_t num_freq_bins,
+                         size_t num_output_channels,
+                         std::complex<float>* const* output) override {
+    float low_pow = 0.f;
+    float high_pow = 0.f;
+    for (size_t i = 0u; i < num_input_channels; ++i) {
+      for (size_t j = 0u; j < kNarrowbandSize; ++j) {
+        float low_mag = std::abs(input[i][j]);
+        low_pow += low_mag * low_mag;
+        float high_mag = std::abs(input[i][j + kNarrowbandSize]);
+        high_pow += high_mag * high_mag;
+      }
+    }
+    low_pow_ += low_pow / (num_input_channels * kFftSize);
+    high_pow_ += high_pow / (num_input_channels * kFftSize);
+  }
+
+ private:
+  std::unique_ptr<LappedTransform> transform_;
+  float window_[kFftSize];
+  float low_pow_;
+  float high_pow_;
+};
+
+float EncodedPowerRatio(AudioEncoder* encoder,
+                        AudioDecoder* decoder,
+                        test::AudioLoop* audio_loop) {
+  // Encode and decode.
+  uint32_t rtp_timestamp = 0u;
+  constexpr size_t kBufferSize = 500;
+  rtc::Buffer encoded(kBufferSize);
+  std::vector<int16_t> decoded(kOutputBlockSizeSamples);
+  std::vector<float> decoded_float(kOutputBlockSizeSamples);
+  AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+  PowerRatioEstimator power_ratio_estimator;
+  for (size_t i = 0; i < 1000; ++i) {
+    encoded.Clear();
+    AudioEncoder::EncodedInfo encoder_info =
+        encoder->Encode(rtp_timestamp, audio_loop->GetNextBlock(), &encoded);
+    rtp_timestamp += kInputBlockSizeSamples;
+    if (encoded.size() > 0) {
+      int decoder_info = decoder->Decode(
+          encoded.data(), encoded.size(), kSampleRateHz,
+          decoded.size() * sizeof(decoded[0]), decoded.data(), &speech_type);
+      if (decoder_info > 0) {
+        S16ToFloat(decoded.data(), decoded.size(), decoded_float.data());
+        power_ratio_estimator.ProcessBlock(decoded_float.data());
+      }
+    }
+  }
+  return power_ratio_estimator.PowerRatio();
+}
+
+}  // namespace
+
+// TODO(ivoc): Remove this test, WebRTC-AdjustOpusBandwidth is obsolete.
+TEST(BandwidthAdaptationTest, BandwidthAdaptationTest) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-AdjustOpusBandwidth/Enabled/");
+
+  constexpr float kMaxNarrowbandRatio = 0.0035f;
+  constexpr float kMinWidebandRatio = 0.01f;
+
+  // Create encoder.
+  AudioEncoderOpusConfig enc_config;
+  enc_config.bitrate_bps = absl::optional<int>(7999);
+  enc_config.num_channels = kNumChannels;
+  constexpr int payload_type = 17;
+  auto encoder = AudioEncoderOpus::MakeAudioEncoder(enc_config, payload_type);
+
+  // Create decoder.
+  AudioDecoderOpus::Config dec_config;
+  dec_config.num_channels = kNumChannels;
+  auto decoder = AudioDecoderOpus::MakeAudioDecoder(dec_config);
+
+  // Open speech file.
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
+  test::AudioLoop audio_loop;
+  EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz());
+  ASSERT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+                              kInputBlockSizeSamples));
+
+  EXPECT_LT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMaxNarrowbandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(9000);
+  EXPECT_LT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMaxNarrowbandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(9001);
+  EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMinWidebandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(8000);
+  EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMinWidebandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(12001);
+  EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMinWidebandRatio);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
new file mode 100644
index 0000000000..e8c131092c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metric.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/time_utils.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::test::GetGlobalMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+int64_t RunComplexityTest(const AudioEncoderOpusConfig& config) {
+  // Create encoder.
+  constexpr int payload_type = 17;
+  const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+  // Open speech file.
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
+  test::AudioLoop audio_loop;
+  constexpr int kSampleRateHz = 48000;
+  EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz());
+  constexpr size_t kMaxLoopLengthSamples =
+      kSampleRateHz * 10;  // 10 second loop.
+  constexpr size_t kInputBlockSizeSamples =
+      10 * kSampleRateHz / 1000;  // 60 ms.
+  EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+                              kInputBlockSizeSamples));
+  // Encode.
+  const int64_t start_time_ms = rtc::TimeMillis();
+  AudioEncoder::EncodedInfo info;
+  rtc::Buffer encoded(500);
+  uint32_t rtp_timestamp = 0u;
+  for (size_t i = 0; i < 10000; ++i) {
+    encoded.Clear();
+    info = encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
+    rtp_timestamp += kInputBlockSizeSamples;
+  }
+  return rtc::TimeMillis() - start_time_ms;
+}
+
+// This test encodes an audio file using Opus twice with different bitrates
+// (~11 kbps and 15.5 kbps). The runtime for each is measured, and the ratio
+// between the two is calculated and tracked. This test explicitly sets the
+// low_rate_complexity to 9. When running on desktop platforms, this is the same
+// as the regular complexity, and the expectation is that the resulting ratio
+// should be less than 100% (since the encoder runs faster at lower bitrates,
+// given a fixed complexity setting). On the other hand, when running on
+// mobiles, the regular complexity is 5, and we expect the resulting ratio to
+// be higher, since we have explicitly asked for a higher complexity setting at
+// the lower rate.
+TEST(AudioEncoderOpusComplexityAdaptationTest, Adaptation_On) {
+  // Create config.
+  AudioEncoderOpusConfig config;
+  // The limit -- including the hysteresis window -- at which the complexity
+  // shuold be increased.
+  config.bitrate_bps = 11000 - 1;
+  config.low_rate_complexity = 9;
+  int64_t runtime_10999bps = RunComplexityTest(config);
+
+  config.bitrate_bps = 15500;
+  int64_t runtime_15500bps = RunComplexityTest(config);
+
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "opus_encoding_complexity_ratio", "adaptation_on",
+      100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+// This test is identical to the one above, but without the complexity
+// adaptation enabled (neither on desktop, nor on mobile). The expectation is
+// that the resulting ratio is less than 100% at all times.
+TEST(AudioEncoderOpusComplexityAdaptationTest, Adaptation_Off) {
+  // Create config.
+  AudioEncoderOpusConfig config;
+  // The limit -- including the hysteresis window -- at which the complexity
+  // shuold be increased (but not in this test since complexity adaptation is
+  // disabled).
+  config.bitrate_bps = 11000 - 1;
+  int64_t runtime_10999bps = RunComplexityTest(config);
+
+  config.bitrate_bps = 15500;
+  int64_t runtime_15500bps = RunComplexityTest(config);
+
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "opus_encoding_complexity_ratio", "adaptation_off",
+      100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
new file mode 100644
index 0000000000..815f26e31c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -0,0 +1,248 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using std::get;
+using std::string;
+using std::tuple;
+using ::testing::TestWithParam;
+
+namespace webrtc {
+
+// Define coding parameter as <channels, bit_rate, filename, extension>.
+typedef tuple<size_t, int, string, string> coding_param;
+typedef struct mode mode;
+
+struct mode {
+  bool fec;
+  uint8_t target_packet_loss_rate;
+};
+
+const int kOpusBlockDurationMs = 20;
+const int kOpusSamplingKhz = 48;
+
+class OpusFecTest : public TestWithParam<coding_param> {
+ protected:
+  OpusFecTest();
+
+  void SetUp() override;
+  void TearDown() override;
+
+  virtual void EncodeABlock();
+
+  virtual void DecodeABlock(bool lost_previous, bool lost_current);
+
+  int block_duration_ms_;
+  int sampling_khz_;
+  size_t block_length_sample_;
+
+  size_t channels_;
+  int bit_rate_;
+
+  size_t data_pointer_;
+  size_t loop_length_samples_;
+  size_t max_bytes_;
+  size_t encoded_bytes_;
+
+  WebRtcOpusEncInst* opus_encoder_;
+  WebRtcOpusDecInst* opus_decoder_;
+
+  string in_filename_;
+
+  std::unique_ptr<int16_t[]> in_data_;
+  std::unique_ptr<int16_t[]> out_data_;
+  std::unique_ptr<uint8_t[]> bit_stream_;
+};
+
+void OpusFecTest::SetUp() {
+  channels_ = get<0>(GetParam());
+  bit_rate_ = get<1>(GetParam());
+  printf("Coding %zu channel signal at %d bps.\n", channels_, bit_rate_);
+
+  in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
+
+  FILE* fp = fopen(in_filename_.c_str(), "rb");
+  ASSERT_FALSE(fp == NULL);
+
+  // Obtain file size.
+  fseek(fp, 0, SEEK_END);
+  loop_length_samples_ = ftell(fp) / sizeof(int16_t);
+  rewind(fp);
+
+  // Allocate memory to contain the whole file.
+  in_data_.reset(
+      new int16_t[loop_length_samples_ + block_length_sample_ * channels_]);
+
+  // Copy the file into the buffer.
+  ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
+            loop_length_samples_);
+  fclose(fp);
+
+  // The audio will be used in a looped manner. To ease the acquisition of an
+  // audio frame that crosses the end of the excerpt, we add an extra block
+  // length of samples to the end of the array, starting over again from the
+  // beginning of the array. Audio frames cross the end of the excerpt always
+  // appear as a continuum of memory.
+  memcpy(&in_data_[loop_length_samples_], &in_data_[0],
+         block_length_sample_ * channels_ * sizeof(int16_t));
+
+  // Maximum number of bytes in output bitstream.
+  max_bytes_ = block_length_sample_ * channels_ * sizeof(int16_t);
+
+  out_data_.reset(new int16_t[2 * block_length_sample_ * channels_]);
+  bit_stream_.reset(new uint8_t[max_bytes_]);
+
+  // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
+  int app = channels_ == 1 ? 0 : 1;
+
+  // Create encoder memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000));
+  EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000));
+  // Set bitrate.
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
+}
+
+void OpusFecTest::TearDown() {
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+OpusFecTest::OpusFecTest()
+    : block_duration_ms_(kOpusBlockDurationMs),
+      sampling_khz_(kOpusSamplingKhz),
+      block_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * sampling_khz_)),
+      data_pointer_(0),
+      max_bytes_(0),
+      encoded_bytes_(0),
+      opus_encoder_(NULL),
+      opus_decoder_(NULL) {}
+
+void OpusFecTest::EncodeABlock() {
+  int value =
+      WebRtcOpus_Encode(opus_encoder_, &in_data_[data_pointer_],
+                        block_length_sample_, max_bytes_, &bit_stream_[0]);
+  EXPECT_GT(value, 0);
+
+  encoded_bytes_ = static_cast<size_t>(value);
+}
+
+void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
+  int16_t audio_type;
+  int value_1 = 0, value_2 = 0;
+
+  if (lost_previous) {
+    // Decode previous frame.
+    if (!lost_current &&
+        WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_) == 1) {
+      value_1 =
+          WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0], encoded_bytes_,
+                               &out_data_[0], &audio_type);
+    } else {
+      // Call decoder PLC.
+      while (value_1 < static_cast<int>(block_length_sample_)) {
+        int ret = WebRtcOpus_Decode(opus_decoder_, NULL, 0, &out_data_[value_1],
+                                    &audio_type);
+        EXPECT_EQ(ret, sampling_khz_ * 10);  // Should return 10 ms of samples.
+        value_1 += ret;
+      }
+    }
+    EXPECT_EQ(static_cast<int>(block_length_sample_), value_1);
+  }
+
+  if (!lost_current) {
+    // Decode current frame.
+    value_2 = WebRtcOpus_Decode(opus_decoder_, &bit_stream_[0], encoded_bytes_,
+                                &out_data_[value_1 * channels_], &audio_type);
+    EXPECT_EQ(static_cast<int>(block_length_sample_), value_2);
+  }
+}
+
+TEST_P(OpusFecTest, RandomPacketLossTest) {
+  const int kDurationMs = 200000;
+  int time_now_ms, fec_frames;
+  int actual_packet_loss_rate;
+  bool lost_current, lost_previous;
+  mode mode_set[3] = {{true, 0}, {false, 0}, {true, 50}};
+
+  lost_current = false;
+  for (int i = 0; i < 3; i++) {
+    if (mode_set[i].fec) {
+      EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
+      EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(
+                       opus_encoder_, mode_set[i].target_packet_loss_rate));
+      printf("FEC is ON, target at packet loss rate %d percent.\n",
+             mode_set[i].target_packet_loss_rate);
+    } else {
+      EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
+      printf("FEC is OFF.\n");
+    }
+    // In this test, we let the target packet loss rate match the actual rate.
+    actual_packet_loss_rate = mode_set[i].target_packet_loss_rate;
+    // Run every mode a certain time.
+    time_now_ms = 0;
+    fec_frames = 0;
+    while (time_now_ms < kDurationMs) {
+      // Encode & decode.
+      EncodeABlock();
+
+      // Check if payload has FEC.
+      int fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
+
+      // If FEC is disabled or the target packet loss rate is set to 0, there
+      // should be no FEC in the bit stream.
+      if (!mode_set[i].fec || mode_set[i].target_packet_loss_rate == 0) {
+        EXPECT_EQ(fec, 0);
+      } else if (fec == 1) {
+        fec_frames++;
+      }
+
+      lost_previous = lost_current;
+      lost_current = rand() < actual_packet_loss_rate * (RAND_MAX / 100);
+      DecodeABlock(lost_previous, lost_current);
+
+      time_now_ms += block_duration_ms_;
+
+      // `data_pointer_` is incremented and wrapped across
+      // `loop_length_samples_`.
+      data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) %
+                      loop_length_samples_;
+    }
+    if (mode_set[i].fec) {
+      printf("%.2f percent frames has FEC.\n",
+             static_cast<float>(fec_frames) * block_duration_ms_ / 2000);
+    }
+  }
+}
+
+const coding_param param_set[] = {
+    std::make_tuple(1,
+                    64000,
+                    string("audio_coding/testfile32kHz"),
+                    string("pcm")),
+    std::make_tuple(1,
+                    32000,
+                    string("audio_coding/testfile32kHz"),
+                    string("pcm")),
+    std::make_tuple(2,
+                    64000,
+                    string("audio_coding/teststereo32kHz"),
+                    string("pcm"))};
+
+// 64 kbps, stereo
+INSTANTIATE_TEST_SUITE_P(AllTest, OpusFecTest, ::testing::ValuesIn(param_set));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h
new file mode 100644
index 0000000000..92c5c354a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+
+#include <stddef.h>
+
+#include "rtc_base/ignore_wundef.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#include "third_party/opus/src/include/opus.h"
+#include "third_party/opus/src/include/opus_multistream.h"
+RTC_POP_IGNORING_WUNDEF()
+
+struct WebRtcOpusEncInst {
+  OpusEncoder* encoder;
+  OpusMSEncoder* multistream_encoder;
+  size_t channels;
+  int in_dtx_mode;
+  bool avoid_noise_pumping_during_dtx;
+  int sample_rate_hz;
+  float smooth_energy_non_active_frames;
+};
+
+struct WebRtcOpusDecInst {
+  OpusDecoder* decoder;
+  OpusMSDecoder* multistream_decoder;
+  int prev_decoded_samples;
+  bool plc_use_prev_decoded_samples;
+  size_t channels;
+  int in_dtx_mode;
+  int sample_rate_hz;
+};
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc
new file mode 100644
index 0000000000..67d8619b34
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc
@@ -0,0 +1,881 @@
+/*
+ *  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 "modules/audio_coding/codecs/opus/opus_interface.h"
+
+#include <cstdlib>
+
+#include <numeric>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+enum {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+  /* Maximum supported frame size in WebRTC is 120 ms. */
+  kWebRtcOpusMaxEncodeFrameSizeMs = 120,
+#else
+  /* Maximum supported frame size in WebRTC is 60 ms. */
+  kWebRtcOpusMaxEncodeFrameSizeMs = 60,
+#endif
+
+  /* The format allows up to 120 ms frames. Since we don't control the other
+   * side, we must allow for packets of that size. NetEq is currently limited
+   * to 60 ms on the receive side. */
+  kWebRtcOpusMaxDecodeFrameSizeMs = 120,
+
+  // Duration of audio that each call to packet loss concealment covers.
+  kWebRtcOpusPlcFrameSizeMs = 10,
+};
+
+constexpr char kPlcUsePrevDecodedSamplesFieldTrial[] =
+    "WebRTC-Audio-OpusPlcUsePrevDecodedSamples";
+
+constexpr char kAvoidNoisePumpingDuringDtxFieldTrial[] =
+    "WebRTC-Audio-OpusAvoidNoisePumpingDuringDtx";
+
+constexpr char kSetSignalVoiceWithDtxFieldTrial[] =
+    "WebRTC-Audio-OpusSetSignalVoiceWithDtx";
+
+static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) {
+  RTC_DCHECK_GT(frame_size_ms, 0);
+  RTC_DCHECK_EQ(frame_size_ms % 10, 0);
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  RTC_DCHECK_EQ(sample_rate_hz % 1000, 0);
+  return frame_size_ms * (sample_rate_hz / 1000);
+}
+
+// Maximum sample count per channel.
+static int MaxFrameSizePerChannel(int sample_rate_hz) {
+  return FrameSizePerChannel(kWebRtcOpusMaxDecodeFrameSizeMs, sample_rate_hz);
+}
+
+// Default sample count per channel.
+static int DefaultFrameSizePerChannel(int sample_rate_hz) {
+  return FrameSizePerChannel(20, sample_rate_hz);
+}
+
+// Returns true if the `encoded` payload corresponds to a refresh DTX packet
+// whose energy is larger than the expected for non activity packets.
+static bool WebRtcOpus_IsHighEnergyRefreshDtxPacket(
+    OpusEncInst* inst,
+    rtc::ArrayView<const int16_t> frame,
+    rtc::ArrayView<const uint8_t> encoded) {
+  if (encoded.size() <= 2) {
+    return false;
+  }
+  int number_frames =
+      frame.size() / DefaultFrameSizePerChannel(inst->sample_rate_hz);
+  if (number_frames > 0 &&
+      WebRtcOpus_PacketHasVoiceActivity(encoded.data(), encoded.size()) == 0) {
+    const float average_frame_energy =
+        std::accumulate(frame.begin(), frame.end(), 0.0f,
+                        [](float a, int32_t b) { return a + b * b; }) /
+        number_frames;
+    if (WebRtcOpus_GetInDtx(inst) == 1 &&
+        average_frame_energy >= inst->smooth_energy_non_active_frames * 0.5f) {
+      // This is a refresh DTX packet as the encoder is in DTX and has
+      // produced a payload > 2 bytes. This refresh packet has a higher energy
+      // than the smooth energy of non activity frames (with a 3 dB negative
+      // margin) and, therefore, it is flagged as a high energy refresh DTX
+      // packet.
+      return true;
+    }
+    // The average energy is tracked in a similar way as the modeling of the
+    // comfort noise in the Silk decoder in Opus
+    // (third_party/opus/src/silk/CNG.c).
+    if (average_frame_energy < inst->smooth_energy_non_active_frames * 0.5f) {
+      inst->smooth_energy_non_active_frames = average_frame_energy;
+    } else {
+      inst->smooth_energy_non_active_frames +=
+          (average_frame_energy - inst->smooth_energy_non_active_frames) *
+          0.25f;
+    }
+  }
+  return false;
+}
+
+int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
+                                 size_t channels,
+                                 int32_t application,
+                                 int sample_rate_hz) {
+  int opus_app;
+  if (!inst)
+    return -1;
+
+  switch (application) {
+    case 0:
+      opus_app = OPUS_APPLICATION_VOIP;
+      break;
+    case 1:
+      opus_app = OPUS_APPLICATION_AUDIO;
+      break;
+    default:
+      return -1;
+  }
+
+  OpusEncInst* state =
+      reinterpret_cast<OpusEncInst*>(calloc(1, sizeof(OpusEncInst)));
+  RTC_DCHECK(state);
+
+  int error;
+  state->encoder = opus_encoder_create(
+      sample_rate_hz, static_cast<int>(channels), opus_app, &error);
+
+  if (error != OPUS_OK || (!state->encoder && !state->multistream_encoder)) {
+    WebRtcOpus_EncoderFree(state);
+    return -1;
+  }
+
+  state->in_dtx_mode = 0;
+  state->channels = channels;
+  state->sample_rate_hz = sample_rate_hz;
+  state->smooth_energy_non_active_frames = 0.0f;
+  state->avoid_noise_pumping_during_dtx =
+      webrtc::field_trial::IsEnabled(kAvoidNoisePumpingDuringDtxFieldTrial);
+
+  *inst = state;
+  return 0;
+}
+
+int16_t WebRtcOpus_MultistreamEncoderCreate(
+    OpusEncInst** inst,
+    size_t channels,
+    int32_t application,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping) {
+  int opus_app;
+  if (!inst)
+    return -1;
+
+  switch (application) {
+    case 0:
+      opus_app = OPUS_APPLICATION_VOIP;
+      break;
+    case 1:
+      opus_app = OPUS_APPLICATION_AUDIO;
+      break;
+    default:
+      return -1;
+  }
+
+  OpusEncInst* state =
+      reinterpret_cast<OpusEncInst*>(calloc(1, sizeof(OpusEncInst)));
+  RTC_DCHECK(state);
+
+  int error;
+  const int sample_rate_hz = 48000;
+  state->multistream_encoder = opus_multistream_encoder_create(
+      sample_rate_hz, channels, streams, coupled_streams, channel_mapping,
+      opus_app, &error);
+
+  if (error != OPUS_OK || (!state->encoder && !state->multistream_encoder)) {
+    WebRtcOpus_EncoderFree(state);
+    return -1;
+  }
+
+  state->in_dtx_mode = 0;
+  state->channels = channels;
+  state->sample_rate_hz = sample_rate_hz;
+  state->smooth_energy_non_active_frames = 0.0f;
+  state->avoid_noise_pumping_during_dtx = false;
+
+  *inst = state;
+  return 0;
+}
+
+int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
+  if (inst) {
+    if (inst->encoder) {
+      opus_encoder_destroy(inst->encoder);
+    } else {
+      opus_multistream_encoder_destroy(inst->multistream_encoder);
+    }
+    free(inst);
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+int WebRtcOpus_Encode(OpusEncInst* inst,
+                      const int16_t* audio_in,
+                      size_t samples,
+                      size_t length_encoded_buffer,
+                      uint8_t* encoded) {
+  int res;
+
+  if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
+    return -1;
+  }
+
+  if (inst->encoder) {
+    res = opus_encode(inst->encoder, (const opus_int16*)audio_in,
+                      static_cast<int>(samples), encoded,
+                      static_cast<opus_int32>(length_encoded_buffer));
+  } else {
+    res = opus_multistream_encode(
+        inst->multistream_encoder, (const opus_int16*)audio_in,
+        static_cast<int>(samples), encoded,
+        static_cast<opus_int32>(length_encoded_buffer));
+  }
+
+  if (res <= 0) {
+    return -1;
+  }
+
+  if (res <= 2) {
+    // Indicates DTX since the packet has nothing but a header. In principle,
+    // there is no need to send this packet. However, we do transmit the first
+    // occurrence to let the decoder know that the encoder enters DTX mode.
+    if (inst->in_dtx_mode) {
+      return 0;
+    } else {
+      inst->in_dtx_mode = 1;
+      return res;
+    }
+  }
+
+  if (inst->avoid_noise_pumping_during_dtx && WebRtcOpus_GetUseDtx(inst) == 1 &&
+      WebRtcOpus_IsHighEnergyRefreshDtxPacket(
+          inst, rtc::MakeArrayView(audio_in, samples),
+          rtc::MakeArrayView(encoded, res))) {
+    // This packet is a high energy refresh DTX packet. For avoiding an increase
+    // of the energy in the DTX region at the decoder, this packet is
+    // substituted by a TOC byte with one empty frame.
+    // The number of frames described in the TOC byte
+    // (https://tools.ietf.org/html/rfc6716#section-3.1) are overwritten to
+    // always indicate one frame (last two bits equal to 0).
+    encoded[0] = encoded[0] & 0b11111100;
+    inst->in_dtx_mode = 1;
+    // The payload is just the TOC byte and has 1 byte as length.
+    return 1;
+  }
+  inst->in_dtx_mode = 0;
+  return res;
+}
+
+#define ENCODER_CTL(inst, vargs)                \
+  (inst->encoder                                \
+       ? opus_encoder_ctl(inst->encoder, vargs) \
+       : opus_multistream_encoder_ctl(inst->multistream_encoder, vargs))
+
+int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_BITRATE(rate));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_PACKET_LOSS_PERC(loss_rate));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz) {
+  opus_int32 set_bandwidth;
+
+  if (!inst)
+    return -1;
+
+  if (frequency_hz <= 8000) {
+    set_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+  } else if (frequency_hz <= 12000) {
+    set_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+  } else if (frequency_hz <= 16000) {
+    set_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+  } else if (frequency_hz <= 24000) {
+    set_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
+  } else {
+    set_bandwidth = OPUS_BANDWIDTH_FULLBAND;
+  }
+  return ENCODER_CTL(inst, OPUS_SET_MAX_BANDWIDTH(set_bandwidth));
+}
+
+int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
+                                      int32_t* result_hz) {
+  if (inst->encoder) {
+    if (opus_encoder_ctl(inst->encoder, OPUS_GET_MAX_BANDWIDTH(result_hz)) ==
+        OPUS_OK) {
+      return 0;
+    }
+    return -1;
+  }
+
+  opus_int32 max_bandwidth;
+  int s;
+  int ret;
+
+  max_bandwidth = 0;
+  ret = OPUS_OK;
+  s = 0;
+  while (ret == OPUS_OK) {
+    OpusEncoder* enc;
+    opus_int32 bandwidth;
+
+    ret = ENCODER_CTL(inst, OPUS_MULTISTREAM_GET_ENCODER_STATE(s, &enc));
+    if (ret == OPUS_BAD_ARG)
+      break;
+    if (ret != OPUS_OK)
+      return -1;
+    if (opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&bandwidth)) != OPUS_OK)
+      return -1;
+
+    if (max_bandwidth != 0 && max_bandwidth != bandwidth)
+      return -1;
+
+    max_bandwidth = bandwidth;
+    s++;
+  }
+  *result_hz = max_bandwidth;
+  return 0;
+}
+
+int16_t WebRtcOpus_EnableFec(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_INBAND_FEC(1));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_DisableFec(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_INBAND_FEC(0));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst) {
+  if (inst) {
+    if (webrtc::field_trial::IsEnabled(kSetSignalVoiceWithDtxFieldTrial)) {
+      int ret = ENCODER_CTL(inst, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
+      if (ret != OPUS_OK) {
+        return ret;
+      }
+    }
+    return ENCODER_CTL(inst, OPUS_SET_DTX(1));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) {
+  if (inst) {
+    if (webrtc::field_trial::IsEnabled(kSetSignalVoiceWithDtxFieldTrial)) {
+      int ret = ENCODER_CTL(inst, OPUS_SET_SIGNAL(OPUS_AUTO));
+      if (ret != OPUS_OK) {
+        return ret;
+      }
+    }
+    return ENCODER_CTL(inst, OPUS_SET_DTX(0));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_GetUseDtx(OpusEncInst* inst) {
+  if (inst) {
+    opus_int32 use_dtx;
+    if (ENCODER_CTL(inst, OPUS_GET_DTX(&use_dtx)) == 0) {
+      return use_dtx;
+    }
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_EnableCbr(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_VBR(0));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_DisableCbr(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_VBR(1));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_COMPLEXITY(complexity));
+  } else {
+    return -1;
+  }
+}
+
+int32_t WebRtcOpus_GetBandwidth(OpusEncInst* inst) {
+  if (!inst) {
+    return -1;
+  }
+  int32_t bandwidth;
+  if (ENCODER_CTL(inst, OPUS_GET_BANDWIDTH(&bandwidth)) == 0) {
+    return bandwidth;
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_BANDWIDTH(bandwidth));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels) {
+  if (!inst)
+    return -1;
+  if (num_channels == 0) {
+    return ENCODER_CTL(inst, OPUS_SET_FORCE_CHANNELS(OPUS_AUTO));
+  } else if (num_channels == 1 || num_channels == 2) {
+    return ENCODER_CTL(inst, OPUS_SET_FORCE_CHANNELS(num_channels));
+  } else {
+    return -1;
+  }
+}
+
+int32_t WebRtcOpus_GetInDtx(OpusEncInst* inst) {
+  if (!inst) {
+    return -1;
+  }
+#ifdef OPUS_GET_IN_DTX
+  int32_t in_dtx;
+  if (ENCODER_CTL(inst, OPUS_GET_IN_DTX(&in_dtx)) == 0) {
+    return in_dtx;
+  }
+#endif
+  return -1;
+}
+
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
+                                 size_t channels,
+                                 int sample_rate_hz) {
+  int error;
+  OpusDecInst* state;
+
+  if (inst != NULL) {
+    // Create Opus decoder state.
+    state = reinterpret_cast<OpusDecInst*>(calloc(1, sizeof(OpusDecInst)));
+    if (state == NULL) {
+      return -1;
+    }
+
+    state->decoder =
+        opus_decoder_create(sample_rate_hz, static_cast<int>(channels), &error);
+    if (error == OPUS_OK && state->decoder) {
+      // Creation of memory all ok.
+      state->channels = channels;
+      state->sample_rate_hz = sample_rate_hz;
+      state->plc_use_prev_decoded_samples =
+          webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
+      if (state->plc_use_prev_decoded_samples) {
+        state->prev_decoded_samples =
+            DefaultFrameSizePerChannel(state->sample_rate_hz);
+      }
+      state->in_dtx_mode = 0;
+      *inst = state;
+      return 0;
+    }
+
+    // If memory allocation was unsuccessful, free the entire state.
+    if (state->decoder) {
+      opus_decoder_destroy(state->decoder);
+    }
+    free(state);
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_MultistreamDecoderCreate(
+    OpusDecInst** inst,
+    size_t channels,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping) {
+  int error;
+  OpusDecInst* state;
+
+  if (inst != NULL) {
+    // Create Opus decoder state.
+    state = reinterpret_cast<OpusDecInst*>(calloc(1, sizeof(OpusDecInst)));
+    if (state == NULL) {
+      return -1;
+    }
+
+    // Create new memory, always at 48000 Hz.
+    state->multistream_decoder = opus_multistream_decoder_create(
+        48000, channels, streams, coupled_streams, channel_mapping, &error);
+
+    if (error == OPUS_OK && state->multistream_decoder) {
+      // Creation of memory all ok.
+      state->channels = channels;
+      state->sample_rate_hz = 48000;
+      state->plc_use_prev_decoded_samples =
+          webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
+      if (state->plc_use_prev_decoded_samples) {
+        state->prev_decoded_samples =
+            DefaultFrameSizePerChannel(state->sample_rate_hz);
+      }
+      state->in_dtx_mode = 0;
+      *inst = state;
+      return 0;
+    }
+
+    // If memory allocation was unsuccessful, free the entire state.
+    opus_multistream_decoder_destroy(state->multistream_decoder);
+    free(state);
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
+  if (inst) {
+    if (inst->decoder) {
+      opus_decoder_destroy(inst->decoder);
+    } else if (inst->multistream_decoder) {
+      opus_multistream_decoder_destroy(inst->multistream_decoder);
+    }
+    free(inst);
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
+  return inst->channels;
+}
+
+void WebRtcOpus_DecoderInit(OpusDecInst* inst) {
+  if (inst->decoder) {
+    opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
+  } else {
+    opus_multistream_decoder_ctl(inst->multistream_decoder, OPUS_RESET_STATE);
+  }
+  inst->in_dtx_mode = 0;
+}
+
+/* For decoder to determine if it is to output speech or comfort noise. */
+static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) {
+  // Audio type becomes comfort noise if `encoded_byte` is 1 and keeps
+  // to be so if the following `encoded_byte` are 0 or 1.
+  if (encoded_bytes == 0 && inst->in_dtx_mode) {
+    return 2;  // Comfort noise.
+  } else if (encoded_bytes == 1 || encoded_bytes == 2) {
+    // TODO(henrik.lundin): There is a slight risk that a 2-byte payload is in
+    // fact a 1-byte TOC with a 1-byte payload. That will be erroneously
+    // interpreted as comfort noise output, but such a payload is probably
+    // faulty anyway.
+
+    // TODO(webrtc:10218): This is wrong for multistream opus. Then are several
+    // single-stream packets glued together with some packet size bytes in
+    // between. See https://tools.ietf.org/html/rfc6716#appendix-B
+    inst->in_dtx_mode = 1;
+    return 2;  // Comfort noise.
+  } else {
+    inst->in_dtx_mode = 0;
+    return 0;  // Speech.
+  }
+}
+
+/* `frame_size` is set to maximum Opus frame size in the normal case, and
+ * is set to the number of samples needed for PLC in case of losses.
+ * It is up to the caller to make sure the value is correct. */
+static int DecodeNative(OpusDecInst* inst,
+                        const uint8_t* encoded,
+                        size_t encoded_bytes,
+                        int frame_size,
+                        int16_t* decoded,
+                        int16_t* audio_type,
+                        int decode_fec) {
+  int res = -1;
+  if (inst->decoder) {
+    res = opus_decode(
+        inst->decoder, encoded, static_cast<opus_int32>(encoded_bytes),
+        reinterpret_cast<opus_int16*>(decoded), frame_size, decode_fec);
+  } else {
+    res = opus_multistream_decode(inst->multistream_decoder, encoded,
+                                  static_cast<opus_int32>(encoded_bytes),
+                                  reinterpret_cast<opus_int16*>(decoded),
+                                  frame_size, decode_fec);
+  }
+
+  if (res <= 0)
+    return -1;
+
+  *audio_type = DetermineAudioType(inst, encoded_bytes);
+
+  return res;
+}
+
+static int DecodePlc(OpusDecInst* inst, int16_t* decoded) {
+  int16_t audio_type = 0;
+  int decoded_samples;
+  int plc_samples =
+      FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
+
+  if (inst->plc_use_prev_decoded_samples) {
+    /* The number of samples we ask for is `number_of_lost_frames` times
+     * `prev_decoded_samples_`. Limit the number of samples to maximum
+     * `MaxFrameSizePerChannel()`. */
+    plc_samples = inst->prev_decoded_samples;
+    const int max_samples_per_channel =
+        MaxFrameSizePerChannel(inst->sample_rate_hz);
+    plc_samples = plc_samples <= max_samples_per_channel
+                      ? plc_samples
+                      : max_samples_per_channel;
+  }
+  decoded_samples =
+      DecodeNative(inst, NULL, 0, plc_samples, decoded, &audio_type, 0);
+  if (decoded_samples < 0) {
+    return -1;
+  }
+
+  return decoded_samples;
+}
+
+int WebRtcOpus_Decode(OpusDecInst* inst,
+                      const uint8_t* encoded,
+                      size_t encoded_bytes,
+                      int16_t* decoded,
+                      int16_t* audio_type) {
+  int decoded_samples;
+
+  if (encoded_bytes == 0) {
+    *audio_type = DetermineAudioType(inst, encoded_bytes);
+    decoded_samples = DecodePlc(inst, decoded);
+  } else {
+    decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
+                                   MaxFrameSizePerChannel(inst->sample_rate_hz),
+                                   decoded, audio_type, 0);
+  }
+  if (decoded_samples < 0) {
+    return -1;
+  }
+
+  if (inst->plc_use_prev_decoded_samples) {
+    /* Update decoded sample memory, to be used by the PLC in case of losses. */
+    inst->prev_decoded_samples = decoded_samples;
+  }
+
+  return decoded_samples;
+}
+
+int WebRtcOpus_DecodeFec(OpusDecInst* inst,
+                         const uint8_t* encoded,
+                         size_t encoded_bytes,
+                         int16_t* decoded,
+                         int16_t* audio_type) {
+  int decoded_samples;
+  int fec_samples;
+
+  if (WebRtcOpus_PacketHasFec(encoded, encoded_bytes) != 1) {
+    return 0;
+  }
+
+  fec_samples =
+      opus_packet_get_samples_per_frame(encoded, inst->sample_rate_hz);
+
+  decoded_samples = DecodeNative(inst, encoded, encoded_bytes, fec_samples,
+                                 decoded, audio_type, 1);
+  if (decoded_samples < 0) {
+    return -1;
+  }
+
+  return decoded_samples;
+}
+
+int WebRtcOpus_DurationEst(OpusDecInst* inst,
+                           const uint8_t* payload,
+                           size_t payload_length_bytes) {
+  if (payload_length_bytes == 0) {
+    // WebRtcOpus_Decode calls PLC when payload length is zero. So we return
+    // PLC duration correspondingly.
+    return WebRtcOpus_PlcDuration(inst);
+  }
+
+  int frames, samples;
+  frames = opus_packet_get_nb_frames(
+      payload, static_cast<opus_int32>(payload_length_bytes));
+  if (frames < 0) {
+    /* Invalid payload data. */
+    return 0;
+  }
+  samples =
+      frames * opus_packet_get_samples_per_frame(payload, inst->sample_rate_hz);
+  if (samples > 120 * inst->sample_rate_hz / 1000) {
+    // More than 120 ms' worth of samples.
+    return 0;
+  }
+  return samples;
+}
+
+int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
+  if (inst->plc_use_prev_decoded_samples) {
+    /* The number of samples we ask for is `number_of_lost_frames` times
+     * `prev_decoded_samples_`. Limit the number of samples to maximum
+     * `MaxFrameSizePerChannel()`. */
+    const int plc_samples = inst->prev_decoded_samples;
+    const int max_samples_per_channel =
+        MaxFrameSizePerChannel(inst->sample_rate_hz);
+    return plc_samples <= max_samples_per_channel ? plc_samples
+                                                  : max_samples_per_channel;
+  }
+  return FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
+}
+
+int WebRtcOpus_FecDurationEst(const uint8_t* payload,
+                              size_t payload_length_bytes,
+                              int sample_rate_hz) {
+  if (WebRtcOpus_PacketHasFec(payload, payload_length_bytes) != 1) {
+    return 0;
+  }
+  const int samples =
+      opus_packet_get_samples_per_frame(payload, sample_rate_hz);
+  const int samples_per_ms = sample_rate_hz / 1000;
+  if (samples < 10 * samples_per_ms || samples > 120 * samples_per_ms) {
+    /* Invalid payload duration. */
+    return 0;
+  }
+  return samples;
+}
+
+int WebRtcOpus_NumSilkFrames(const uint8_t* payload) {
+  // For computing the payload length in ms, the sample rate is not important
+  // since it cancels out. We use 48 kHz, but any valid sample rate would work.
+  int payload_length_ms =
+      opus_packet_get_samples_per_frame(payload, 48000) / 48;
+  if (payload_length_ms < 10)
+    payload_length_ms = 10;
+
+  int silk_frames;
+  switch (payload_length_ms) {
+    case 10:
+    case 20:
+      silk_frames = 1;
+      break;
+    case 40:
+      silk_frames = 2;
+      break;
+    case 60:
+      silk_frames = 3;
+      break;
+    default:
+      return 0;  // It is actually even an invalid packet.
+  }
+  return silk_frames;
+}
+
+// This method is based on Definition of the Opus Audio Codec
+// (https://tools.ietf.org/html/rfc6716). Basically, this method is based on
+// parsing the LP layer of an Opus packet, particularly the LBRR flag.
+int WebRtcOpus_PacketHasFec(const uint8_t* payload,
+                            size_t payload_length_bytes) {
+  if (payload == NULL || payload_length_bytes == 0)
+    return 0;
+
+  // In CELT_ONLY mode, packets should not have FEC.
+  if (payload[0] & 0x80)
+    return 0;
+
+  int silk_frames = WebRtcOpus_NumSilkFrames(payload);
+  if (silk_frames == 0)
+    return 0;  // Not valid.
+
+  const int channels = opus_packet_get_nb_channels(payload);
+  RTC_DCHECK(channels == 1 || channels == 2);
+
+  // Max number of frames in an Opus packet is 48.
+  opus_int16 frame_sizes[48];
+  const unsigned char* frame_data[48];
+
+  // Parse packet to get the frames. But we only care about the first frame,
+  // since we can only decode the FEC from the first one.
+  if (opus_packet_parse(payload, static_cast<opus_int32>(payload_length_bytes),
+                        NULL, frame_data, frame_sizes, NULL) < 0) {
+    return 0;
+  }
+
+  if (frame_sizes[0] < 1) {
+    return 0;
+  }
+
+  // A frame starts with the LP layer. The LP layer begins with two to eight
+  // header bits.These consist of one VAD bit per SILK frame (up to 3),
+  // followed by a single flag indicating the presence of LBRR frames.
+  // For a stereo packet, these first flags correspond to the mid channel, and
+  // a second set of flags is included for the side channel. Because these are
+  // the first symbols decoded by the range coder and because they are coded
+  // as binary values with uniform probability, they can be extracted directly
+  // from the most significant bits of the first byte of compressed data.
+  for (int n = 0; n < channels; n++) {
+    // The LBRR bit for channel 1 is on the (`silk_frames` + 1)-th bit, and
+    // that of channel 2 is on the |(`silk_frames` + 1) * 2 + 1|-th bit.
+    if (frame_data[0][0] & (0x80 >> ((n + 1) * (silk_frames + 1) - 1)))
+      return 1;
+  }
+
+  return 0;
+}
+
+int WebRtcOpus_PacketHasVoiceActivity(const uint8_t* payload,
+                                      size_t payload_length_bytes) {
+  if (payload == NULL || payload_length_bytes == 0)
+    return 0;
+
+  // In CELT_ONLY mode we can not determine whether there is VAD.
+  if (payload[0] & 0x80)
+    return -1;
+
+  int silk_frames = WebRtcOpus_NumSilkFrames(payload);
+  if (silk_frames == 0)
+    return -1;
+
+  const int channels = opus_packet_get_nb_channels(payload);
+  RTC_DCHECK(channels == 1 || channels == 2);
+
+  // Max number of frames in an Opus packet is 48.
+  opus_int16 frame_sizes[48];
+  const unsigned char* frame_data[48];
+
+  // Parse packet to get the frames.
+  int frames =
+      opus_packet_parse(payload, static_cast<opus_int32>(payload_length_bytes),
+                        NULL, frame_data, frame_sizes, NULL);
+  if (frames < 0)
+    return -1;
+
+  // Iterate over all Opus frames which may contain multiple SILK frames.
+  for (int frame = 0; frame < frames; frame++) {
+    if (frame_sizes[frame] < 1) {
+      continue;
+    }
+    if (frame_data[frame][0] >> (8 - silk_frames))
+      return 1;
+    if (channels == 2 &&
+        (frame_data[frame][0] << (silk_frames + 1)) >> (8 - silk_frames))
+      return 1;
+  }
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.h
new file mode 100644
index 0000000000..89159ce1c0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.h
@@ -0,0 +1,547 @@
+/*
+ *  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 MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+// Opaque wrapper types for the codec state.
+typedef struct WebRtcOpusEncInst OpusEncInst;
+typedef struct WebRtcOpusDecInst OpusDecInst;
+
+/****************************************************************************
+ * WebRtcOpus_EncoderCreate(...)
+ *
+ * This function creates an Opus encoder that encodes mono or stereo.
+ *
+ * Input:
+ *      - channels           : number of channels; 1 or 2.
+ *      - application        : 0 - VOIP applications.
+ *                                 Favor speech intelligibility.
+ *                             1 - Audio applications.
+ *                                 Favor faithfulness to the original input.
+ *      - sample_rate_hz     : sample rate of input audio
+ *
+ * Output:
+ *      - inst               : a pointer to Encoder context that is created
+ *                             if success.
+ *
+ * Return value              : 0 - Success
+ *                            -1 - Error
+ */
+int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
+                                 size_t channels,
+                                 int32_t application,
+                                 int sample_rate_hz);
+
+/****************************************************************************
+ * WebRtcOpus_MultistreamEncoderCreate(...)
+ *
+ * This function creates an Opus encoder with any supported channel count.
+ *
+ * Input:
+ *      - channels           : number of channels in the input of the encoder.
+ *      - application        : 0 - VOIP applications.
+ *                                 Favor speech intelligibility.
+ *                             1 - Audio applications.
+ *                                 Favor faithfulness to the original input.
+ *      - streams            : number of streams, as described in RFC 7845.
+ *      - coupled_streams    : number of coupled streams, as described in
+ *                             RFC 7845.
+ *      - channel_mapping    : the channel mapping; pointer to array of
+ *                             `channel` bytes, as described in RFC 7845.
+ *
+ * Output:
+ *      - inst               : a pointer to Encoder context that is created
+ *                             if success.
+ *
+ * Return value              : 0 - Success
+ *                            -1 - Error
+ */
+int16_t WebRtcOpus_MultistreamEncoderCreate(
+    OpusEncInst** inst,
+    size_t channels,
+    int32_t application,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping);
+
+int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_Encode(...)
+ *
+ * This function encodes audio as a series of Opus frames and inserts
+ * it into a packet. Input buffer can be any length.
+ *
+ * Input:
+ *      - inst                  : Encoder context
+ *      - audio_in              : Input speech data buffer
+ *      - samples               : Samples per channel in audio_in
+ *      - length_encoded_buffer : Output buffer size
+ *
+ * Output:
+ *      - encoded               : Output compressed data buffer
+ *
+ * Return value                 : >=0 - Length (in bytes) of coded data
+ *                                -1 - Error
+ */
+int WebRtcOpus_Encode(OpusEncInst* inst,
+                      const int16_t* audio_in,
+                      size_t samples,
+                      size_t length_encoded_buffer,
+                      uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcOpus_SetBitRate(...)
+ *
+ * This function adjusts the target bitrate of the encoder.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - rate               : New target bitrate
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate);
+
+/****************************************************************************
+ * WebRtcOpus_SetPacketLossRate(...)
+ *
+ * This function configures the encoder's expected packet loss percentage.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - loss_rate          : loss percentage in the range 0-100, inclusive.
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate);
+
+/****************************************************************************
+ * WebRtcOpus_SetMaxPlaybackRate(...)
+ *
+ * Configures the maximum playback rate for encoding. Due to hardware
+ * limitations, the receiver may render audio up to a playback rate. Opus
+ * encoder can use this information to optimize for network usage and encoding
+ * complexity. This will affect the audio bandwidth in the coded audio. However,
+ * the input/output sample rate is not affected.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - frequency_hz       : Maximum playback rate in Hz.
+ *                             This parameter can take any value. The relation
+ *                             between the value and the Opus internal mode is
+ *                             as following:
+ *                             frequency_hz <= 8000           narrow band
+ *                             8000 < frequency_hz <= 12000   medium band
+ *                             12000 < frequency_hz <= 16000  wide band
+ *                             16000 < frequency_hz <= 24000  super wide band
+ *                             frequency_hz > 24000           full band
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz);
+
+/****************************************************************************
+ * WebRtcOpus_GetMaxPlaybackRate(...)
+ *
+ * Queries the maximum playback rate for encoding. If different single-stream
+ * encoders have different maximum playback rates, this function fails.
+ *
+ * Input:
+ *      - inst               : Encoder context.
+ * Output:
+ *      - result_hz          : The maximum playback rate in Hz.
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
+                                      int32_t* result_hz);
+
+/* TODO(minyue): Check whether an API to check the FEC and the packet loss rate
+ * is needed. It might not be very useful since there are not many use cases and
+ * the caller can always maintain the states. */
+
+/****************************************************************************
+ * WebRtcOpus_EnableFec()
+ *
+ * This function enables FEC for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_EnableFec(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DisableFec()
+ *
+ * This function disables FEC for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_DisableFec(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_EnableDtx()
+ *
+ * This function enables Opus internal DTX for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DisableDtx()
+ *
+ * This function disables Opus internal DTX for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_GetUseDtx()
+ *
+ * This function gets the DTX configuration used for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Encoder does not use DTX.
+ *                              1 - Encoder uses DTX.
+ *                             -1 - Error.
+ */
+int16_t WebRtcOpus_GetUseDtx(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_EnableCbr()
+ *
+ * This function enables CBR for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_EnableCbr(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DisableCbr()
+ *
+ * This function disables CBR for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_DisableCbr(OpusEncInst* inst);
+
+/*
+ * WebRtcOpus_SetComplexity(...)
+ *
+ * This function adjusts the computational complexity. The effect is the same as
+ * calling the complexity setting of Opus as an Opus encoder related CTL.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - complexity         : New target complexity (0-10, inclusive)
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity);
+
+/*
+ * WebRtcOpus_GetBandwidth(...)
+ *
+ * This function returns the current bandwidth.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  Bandwidth - Success
+ *                             -1 - Error
+ */
+int32_t WebRtcOpus_GetBandwidth(OpusEncInst* inst);
+
+/*
+ * WebRtcOpus_SetBandwidth(...)
+ *
+ * By default Opus decides which bandwidth to encode the signal in depending on
+ * the the bitrate. This function overrules the previous setting and forces the
+ * encoder to encode in narrowband/wideband/fullband/etc.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - bandwidth          : New target bandwidth. Valid values are:
+ *                             OPUS_BANDWIDTH_NARROWBAND
+ *                             OPUS_BANDWIDTH_MEDIUMBAND
+ *                             OPUS_BANDWIDTH_WIDEBAND
+ *                             OPUS_BANDWIDTH_SUPERWIDEBAND
+ *                             OPUS_BANDWIDTH_FULLBAND
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth);
+
+/*
+ * WebRtcOpus_GetInDtx(...)
+ *
+ * Gets the DTX state of the encoder.
+ *
+ * Input:
+ *      - inst   : Encoder context
+ *
+ * Return value  : -1 - Error.
+ *                 1  - Last encoded frame was comfort noise update during DTX.
+ *                 0  - Last encoded frame was encoded with encoder not in DTX.
+ */
+int32_t WebRtcOpus_GetInDtx(OpusEncInst* inst);
+
+/*
+ * WebRtcOpus_SetForceChannels(...)
+ *
+ * If the encoder is initialized as a stereo encoder, Opus will by default
+ * decide whether to encode in mono or stereo based on the bitrate. This
+ * function overrules the previous setting, and forces the encoder to encode
+ * in auto/mono/stereo.
+ *
+ * If the Encoder is initialized as a mono encoder, and one tries to force
+ * stereo, the function will return an error.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - num_channels       : 0 - Not forced
+ *                             1 - Mono
+ *                             2 - Stereo
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels);
+
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
+                                 size_t channels,
+                                 int sample_rate_hz);
+
+/****************************************************************************
+ * WebRtcOpus_MultistreamDecoderCreate(...)
+ *
+ * This function creates an Opus decoder with any supported channel count.
+ *
+ * Input:
+ *      - channels           : number of output channels that the decoder
+ *                             will produce.
+ *      - streams            : number of encoded streams, as described in
+ *                             RFC 7845.
+ *      - coupled_streams    : number of coupled streams, as described in
+ *                             RFC 7845.
+ *      - channel_mapping    : the channel mapping; pointer to array of
+ *                             `channel` bytes, as described in RFC 7845.
+ *
+ * Output:
+ *      - inst               : a pointer to a Decoder context that is created
+ *                             if success.
+ *
+ * Return value              : 0 - Success
+ *                            -1 - Error
+ */
+int16_t WebRtcOpus_MultistreamDecoderCreate(
+    OpusDecInst** inst,
+    size_t channels,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping);
+
+int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DecoderChannels(...)
+ *
+ * This function returns the number of channels created for Opus decoder.
+ */
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DecoderInit(...)
+ *
+ * This function resets state of the decoder.
+ *
+ * Input:
+ *      - inst               : Decoder context
+ */
+void WebRtcOpus_DecoderInit(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_Decode(...)
+ *
+ * This function decodes an Opus packet into one or more audio frames at the
+ * ACM interface's sampling rate (32 kHz).
+ *
+ * Input:
+ *      - inst               : Decoder context
+ *      - encoded            : Encoded data
+ *      - encoded_bytes      : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - audio_type         : 1 normal, 2 CNG (for Opus it should
+ *                             always return 1 since we're not using Opus's
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples per channel in decoded vector
+ *                             -1 - Error
+ */
+int WebRtcOpus_Decode(OpusDecInst* inst,
+                      const uint8_t* encoded,
+                      size_t encoded_bytes,
+                      int16_t* decoded,
+                      int16_t* audio_type);
+
+/****************************************************************************
+ * WebRtcOpus_DecodeFec(...)
+ *
+ * This function decodes the FEC data from an Opus packet into one or more audio
+ * frames at the ACM interface's sampling rate (32 kHz).
+ *
+ * Input:
+ *      - inst               : Decoder context
+ *      - encoded            : Encoded data
+ *      - encoded_bytes      : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector (previous frame)
+ *
+ * Return value              : >0 - Samples per channel in decoded vector
+ *                              0 - No FEC data in the packet
+ *                             -1 - Error
+ */
+int WebRtcOpus_DecodeFec(OpusDecInst* inst,
+                         const uint8_t* encoded,
+                         size_t encoded_bytes,
+                         int16_t* decoded,
+                         int16_t* audio_type);
+
+/****************************************************************************
+ * WebRtcOpus_DurationEst(...)
+ *
+ * This function calculates the duration of an opus packet.
+ * Input:
+ *        - inst                 : Decoder context
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *
+ * Return value                  : The duration of the packet, in samples per
+ *                                 channel.
+ */
+int WebRtcOpus_DurationEst(OpusDecInst* inst,
+                           const uint8_t* payload,
+                           size_t payload_length_bytes);
+
+/****************************************************************************
+ * WebRtcOpus_PlcDuration(...)
+ *
+ * This function calculates the duration of a frame returned by packet loss
+ * concealment (PLC).
+ *
+ * Input:
+ *        - inst                 : Decoder context
+ *
+ * Return value                  : The duration of a frame returned by PLC, in
+ *                                 samples per channel.
+ */
+int WebRtcOpus_PlcDuration(OpusDecInst* inst);
+
+/* TODO(minyue): Check whether it is needed to add a decoder context to the
+ * arguments, like WebRtcOpus_DurationEst(...). In fact, the packet itself tells
+ * the duration. The decoder context in WebRtcOpus_DurationEst(...) is not used.
+ * So it may be advisable to remove it from WebRtcOpus_DurationEst(...). */
+
+/****************************************************************************
+ * WebRtcOpus_FecDurationEst(...)
+ *
+ * This function calculates the duration of the FEC data within an opus packet.
+ * Input:
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *        - sample_rate_hz       : Sample rate of output audio
+ *
+ * Return value                  : >0 - The duration of the FEC data in the
+ *                                 packet in samples per channel.
+ *                                  0 - No FEC data in the packet.
+ */
+int WebRtcOpus_FecDurationEst(const uint8_t* payload,
+                              size_t payload_length_bytes,
+                              int sample_rate_hz);
+
+/****************************************************************************
+ * WebRtcOpus_PacketHasFec(...)
+ *
+ * This function detects if an opus packet has FEC.
+ * Input:
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *
+ * Return value                  : 0 - the packet does NOT contain FEC.
+ *                                 1 - the packet contains FEC.
+ */
+int WebRtcOpus_PacketHasFec(const uint8_t* payload,
+                            size_t payload_length_bytes);
+
+/****************************************************************************
+ * WebRtcOpus_PacketHasVoiceActivity(...)
+ *
+ * This function returns the SILK VAD information encoded in the opus packet.
+ * For CELT-only packets that do not have VAD information, it returns -1.
+ * Input:
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *
+ * Return value                  : 0 - no frame had the VAD flag set.
+ *                                 1 - at least one frame had the VAD flag set.
+ *                                -1 - VAD status could not be determined.
+ */
+int WebRtcOpus_PacketHasVoiceActivity(const uint8_t* payload,
+                                      size_t payload_length_bytes);
+
+#ifdef __cplusplus
+}  // extern "C"
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
new file mode 100644
index 0000000000..4477e8a5f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
+
+using ::std::string;
+
+namespace webrtc {
+
+static const int kOpusBlockDurationMs = 20;
+static const int kOpusSamplingKhz = 48;
+
+class OpusSpeedTest : public AudioCodecSpeedTest {
+ protected:
+  OpusSpeedTest();
+  void SetUp() override;
+  void TearDown() override;
+  float EncodeABlock(int16_t* in_data,
+                     uint8_t* bit_stream,
+                     size_t max_bytes,
+                     size_t* encoded_bytes) override;
+  float DecodeABlock(const uint8_t* bit_stream,
+                     size_t encoded_bytes,
+                     int16_t* out_data) override;
+  WebRtcOpusEncInst* opus_encoder_;
+  WebRtcOpusDecInst* opus_decoder_;
+};
+
+OpusSpeedTest::OpusSpeedTest()
+    : AudioCodecSpeedTest(kOpusBlockDurationMs,
+                          kOpusSamplingKhz,
+                          kOpusSamplingKhz),
+      opus_encoder_(NULL),
+      opus_decoder_(NULL) {}
+
+void OpusSpeedTest::SetUp() {
+  AudioCodecSpeedTest::SetUp();
+  // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
+  int app = channels_ == 1 ? 0 : 1;
+  /* Create encoder memory. */
+  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000));
+  EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000));
+  /* Set bitrate. */
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
+}
+
+void OpusSpeedTest::TearDown() {
+  AudioCodecSpeedTest::TearDown();
+  /* Free memory. */
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+float OpusSpeedTest::EncodeABlock(int16_t* in_data,
+                                  uint8_t* bit_stream,
+                                  size_t max_bytes,
+                                  size_t* encoded_bytes) {
+  clock_t clocks = clock();
+  int value = WebRtcOpus_Encode(opus_encoder_, in_data, input_length_sample_,
+                                max_bytes, bit_stream);
+  clocks = clock() - clocks;
+  EXPECT_GT(value, 0);
+  *encoded_bytes = static_cast<size_t>(value);
+  return 1000.0 * clocks / CLOCKS_PER_SEC;
+}
+
+float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
+                                  size_t encoded_bytes,
+                                  int16_t* out_data) {
+  int value;
+  int16_t audio_type;
+  clock_t clocks = clock();
+  value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data,
+                            &audio_type);
+  clocks = clock() - clocks;
+  EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
+  return 1000.0 * clocks / CLOCKS_PER_SEC;
+}
+
+/* Test audio length in second. */
+constexpr size_t kDurationSec = 400;
+
+#define ADD_TEST(complexity)                                           \
+  TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) {           \
+    /* Set complexity. */                                              \
+    printf("Setting complexity to %d ...\n", complexity);              \
+    EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
+    EncodeDecode(kDurationSec);                                        \
+  }
+
+ADD_TEST(10)
+ADD_TEST(9)
+ADD_TEST(8)
+ADD_TEST(7)
+ADD_TEST(6)
+ADD_TEST(5)
+ADD_TEST(4)
+ADD_TEST(3)
+ADD_TEST(2)
+ADD_TEST(1)
+ADD_TEST(0)
+
+#define ADD_BANDWIDTH_TEST(bandwidth)                                \
+  TEST_P(OpusSpeedTest, OpusSetBandwidthTest##bandwidth) {           \
+    /* Set bandwidth. */                                             \
+    printf("Setting bandwidth to %d ...\n", bandwidth);              \
+    EXPECT_EQ(0, WebRtcOpus_SetBandwidth(opus_encoder_, bandwidth)); \
+    EncodeDecode(kDurationSec);                                      \
+  }
+
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_NARROWBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_MEDIUMBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_WIDEBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_SUPERWIDEBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_FULLBAND)
+
+// List all test cases: (channel, bit rat, filename, extension).
+const coding_param param_set[] = {
+    std::make_tuple(1,
+                    64000,
+                    string("audio_coding/speech_mono_32_48kHz"),
+                    string("pcm"),
+                    true),
+    std::make_tuple(1,
+                    32000,
+                    string("audio_coding/speech_mono_32_48kHz"),
+                    string("pcm"),
+                    true),
+    std::make_tuple(2,
+                    64000,
+                    string("audio_coding/music_stereo_48kHz"),
+                    string("pcm"),
+                    true)};
+
+INSTANTIATE_TEST_SUITE_P(AllTest,
+                         OpusSpeedTest,
+                         ::testing::ValuesIn(param_set));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
new file mode 100644
index 0000000000..4a9156ad58
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -0,0 +1,979 @@
+/*
+ *  Copyright (c) 2013 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 <memory>
+#include <string>
+
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace {
+// Equivalent to SDP params
+// {{"channel_mapping", "0,1,2,3"}, {"coupled_streams", "2"}}.
+constexpr unsigned char kQuadChannelMapping[] = {0, 1, 2, 3};
+constexpr int kQuadTotalStreams = 2;
+constexpr int kQuadCoupledStreams = 2;
+
+constexpr unsigned char kStereoChannelMapping[] = {0, 1};
+constexpr int kStereoTotalStreams = 1;
+constexpr int kStereoCoupledStreams = 1;
+
+constexpr unsigned char kMonoChannelMapping[] = {0};
+constexpr int kMonoTotalStreams = 1;
+constexpr int kMonoCoupledStreams = 0;
+
+void CreateSingleOrMultiStreamEncoder(WebRtcOpusEncInst** opus_encoder,
+                                      int channels,
+                                      int application,
+                                      bool use_multistream,
+                                      int encoder_sample_rate_hz) {
+  EXPECT_TRUE(channels == 1 || channels == 2 || use_multistream);
+  if (use_multistream) {
+    EXPECT_EQ(encoder_sample_rate_hz, 48000);
+    if (channels == 1) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
+                       opus_encoder, channels, application, kMonoTotalStreams,
+                       kMonoCoupledStreams, kMonoChannelMapping));
+    } else if (channels == 2) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
+                       opus_encoder, channels, application, kStereoTotalStreams,
+                       kStereoCoupledStreams, kStereoChannelMapping));
+    } else if (channels == 4) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
+                       opus_encoder, channels, application, kQuadTotalStreams,
+                       kQuadCoupledStreams, kQuadChannelMapping));
+    } else {
+      EXPECT_TRUE(false) << channels;
+    }
+  } else {
+    EXPECT_EQ(0, WebRtcOpus_EncoderCreate(opus_encoder, channels, application,
+                                          encoder_sample_rate_hz));
+  }
+}
+
+void CreateSingleOrMultiStreamDecoder(WebRtcOpusDecInst** opus_decoder,
+                                      int channels,
+                                      bool use_multistream,
+                                      int decoder_sample_rate_hz) {
+  EXPECT_TRUE(channels == 1 || channels == 2 || use_multistream);
+  if (use_multistream) {
+    EXPECT_EQ(decoder_sample_rate_hz, 48000);
+    if (channels == 1) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
+                       opus_decoder, channels, kMonoTotalStreams,
+                       kMonoCoupledStreams, kMonoChannelMapping));
+    } else if (channels == 2) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
+                       opus_decoder, channels, kStereoTotalStreams,
+                       kStereoCoupledStreams, kStereoChannelMapping));
+    } else if (channels == 4) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
+                       opus_decoder, channels, kQuadTotalStreams,
+                       kQuadCoupledStreams, kQuadChannelMapping));
+    } else {
+      EXPECT_TRUE(false) << channels;
+    }
+  } else {
+    EXPECT_EQ(0, WebRtcOpus_DecoderCreate(opus_decoder, channels,
+                                          decoder_sample_rate_hz));
+  }
+}
+
+int SamplesPerChannel(int sample_rate_hz, int duration_ms) {
+  const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz, 1000);
+  return samples_per_ms * duration_ms;
+}
+
+using test::AudioLoop;
+using ::testing::Combine;
+using ::testing::TestWithParam;
+using ::testing::Values;
+
+// Maximum number of bytes in output bitstream.
+const size_t kMaxBytes = 2000;
+
+class OpusTest
+    : public TestWithParam<::testing::tuple<size_t, int, bool, int, int>> {
+ protected:
+  OpusTest() = default;
+
+  void TestDtxEffect(bool dtx, int block_length_ms);
+
+  void TestCbrEffect(bool dtx, int block_length_ms);
+
+  // Prepare `speech_data_` for encoding, read from a hard-coded file.
+  // After preparation, `speech_data_.GetNextBlock()` returns a pointer to a
+  // block of `block_length_ms` milliseconds. The data is looped every
+  // `loop_length_ms` milliseconds.
+  void PrepareSpeechData(int block_length_ms, int loop_length_ms);
+
+  int EncodeDecode(WebRtcOpusEncInst* encoder,
+                   rtc::ArrayView<const int16_t> input_audio,
+                   WebRtcOpusDecInst* decoder,
+                   int16_t* output_audio,
+                   int16_t* audio_type);
+
+  void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
+                          opus_int32 expect,
+                          int32_t set);
+
+  void CheckAudioBounded(const int16_t* audio,
+                         size_t samples,
+                         size_t channels,
+                         uint16_t bound) const;
+
+  WebRtcOpusEncInst* opus_encoder_ = nullptr;
+  WebRtcOpusDecInst* opus_decoder_ = nullptr;
+  AudioLoop speech_data_;
+  uint8_t bitstream_[kMaxBytes];
+  size_t encoded_bytes_ = 0;
+  const size_t channels_{std::get<0>(GetParam())};
+  const int application_{std::get<1>(GetParam())};
+  const bool use_multistream_{std::get<2>(GetParam())};
+  const int encoder_sample_rate_hz_{std::get<3>(GetParam())};
+  const int decoder_sample_rate_hz_{std::get<4>(GetParam())};
+};
+
+}  // namespace
+
+// Singlestream: Try all combinations.
+INSTANTIATE_TEST_SUITE_P(Singlestream,
+                         OpusTest,
+                         testing::Combine(testing::Values(1, 2),
+                                          testing::Values(0, 1),
+                                          testing::Values(false),
+                                          testing::Values(16000, 48000),
+                                          testing::Values(16000, 48000)));
+
+// Multistream: Some representative cases (only 48 kHz for now).
+INSTANTIATE_TEST_SUITE_P(
+    Multistream,
+    OpusTest,
+    testing::Values(std::make_tuple(1, 0, true, 48000, 48000),
+                    std::make_tuple(2, 1, true, 48000, 48000),
+                    std::make_tuple(4, 0, true, 48000, 48000),
+                    std::make_tuple(4, 1, true, 48000, 48000)));
+
+void OpusTest::PrepareSpeechData(int block_length_ms, int loop_length_ms) {
+  std::map<int, std::string> channel_to_basename = {
+      {1, "audio_coding/testfile32kHz"},
+      {2, "audio_coding/teststereo32kHz"},
+      {4, "audio_coding/speech_4_channels_48k_one_second"}};
+  std::map<int, std::string> channel_to_suffix = {
+      {1, "pcm"}, {2, "pcm"}, {4, "wav"}};
+  const std::string file_name = webrtc::test::ResourcePath(
+      channel_to_basename[channels_], channel_to_suffix[channels_]);
+  if (loop_length_ms < block_length_ms) {
+    loop_length_ms = block_length_ms;
+  }
+  const int sample_rate_khz =
+      rtc::CheckedDivExact(encoder_sample_rate_hz_, 1000);
+  EXPECT_TRUE(speech_data_.Init(file_name,
+                                loop_length_ms * sample_rate_khz * channels_,
+                                block_length_ms * sample_rate_khz * channels_));
+}
+
+void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
+                                  opus_int32 expect,
+                                  int32_t set) {
+  opus_int32 bandwidth;
+  EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, set));
+  EXPECT_EQ(0, WebRtcOpus_GetMaxPlaybackRate(opus_encoder_, &bandwidth));
+  EXPECT_EQ(expect, bandwidth);
+}
+
+void OpusTest::CheckAudioBounded(const int16_t* audio,
+                                 size_t samples,
+                                 size_t channels,
+                                 uint16_t bound) const {
+  for (size_t i = 0; i < samples; ++i) {
+    for (size_t c = 0; c < channels; ++c) {
+      ASSERT_GE(audio[i * channels + c], -bound);
+      ASSERT_LE(audio[i * channels + c], bound);
+    }
+  }
+}
+
+int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
+                           rtc::ArrayView<const int16_t> input_audio,
+                           WebRtcOpusDecInst* decoder,
+                           int16_t* output_audio,
+                           int16_t* audio_type) {
+  const int input_samples_per_channel =
+      rtc::CheckedDivExact(input_audio.size(), channels_);
+  int encoded_bytes_int =
+      WebRtcOpus_Encode(encoder, input_audio.data(), input_samples_per_channel,
+                        kMaxBytes, bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
+  encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
+  if (encoded_bytes_ != 0) {
+    int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_);
+    int act_len = WebRtcOpus_Decode(decoder, bitstream_, encoded_bytes_,
+                                    output_audio, audio_type);
+    EXPECT_EQ(est_len, act_len);
+    return act_len;
+  } else {
+    int total_dtx_len = 0;
+    const int output_samples_per_channel = input_samples_per_channel *
+                                           decoder_sample_rate_hz_ /
+                                           encoder_sample_rate_hz_;
+    while (total_dtx_len < output_samples_per_channel) {
+      int est_len = WebRtcOpus_DurationEst(decoder, NULL, 0);
+      int act_len = WebRtcOpus_Decode(decoder, NULL, 0,
+                                      &output_audio[total_dtx_len * channels_],
+                                      audio_type);
+      EXPECT_EQ(est_len, act_len);
+      total_dtx_len += act_len;
+    }
+    return total_dtx_len;
+  }
+}
+
+// Test if encoder/decoder can enter DTX mode properly and do not enter DTX when
+// they should not. This test is signal dependent.
+void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
+  PrepareSpeechData(block_length_ms, 2000);
+  const size_t input_samples =
+      rtc::CheckedDivExact(encoder_sample_rate_hz_, 1000) * block_length_ms;
+  const size_t output_samples =
+      rtc::CheckedDivExact(decoder_sample_rate_hz_, 1000) * block_length_ms;
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Set input audio as silence.
+  std::vector<int16_t> silence(input_samples * channels_, 0);
+
+  // Setting DTX.
+  EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_)
+                   : WebRtcOpus_DisableDtx(opus_encoder_));
+
+  int16_t audio_type;
+  int16_t* output_data_decode = new int16_t[output_samples * channels_];
+
+  for (int i = 0; i < 100; ++i) {
+    EXPECT_EQ(output_samples,
+              static_cast<size_t>(EncodeDecode(
+                  opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+                  output_data_decode, &audio_type)));
+    // If not DTX, it should never enter DTX mode. If DTX, we do not care since
+    // whether it enters DTX depends on the signal type.
+    if (!dtx) {
+      EXPECT_GT(encoded_bytes_, 1U);
+      EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(0, audio_type);  // Speech.
+    }
+  }
+
+  // We input some silent segments. In DTX mode, the encoder will stop sending.
+  // However, DTX may happen after a while.
+  for (int i = 0; i < 30; ++i) {
+    EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                  opus_encoder_, silence, opus_decoder_,
+                                  output_data_decode, &audio_type)));
+    if (!dtx) {
+      EXPECT_GT(encoded_bytes_, 1U);
+      EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(0, audio_type);  // Speech.
+    } else if (encoded_bytes_ == 1) {
+      EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(2, audio_type);  // Comfort noise.
+      break;
+    }
+  }
+
+  // When Opus is in DTX, it wakes up in a regular basis. It sends two packets,
+  // one with an arbitrary size and the other of 1-byte, then stops sending for
+  // a certain number of frames.
+
+  // `max_dtx_frames` is the maximum number of frames Opus can stay in DTX.
+  // TODO(kwiberg): Why does this number depend on the encoding sample rate?
+  const int max_dtx_frames =
+      (encoder_sample_rate_hz_ == 16000 ? 800 : 400) / block_length_ms + 1;
+
+  // We run `kRunTimeMs` milliseconds of pure silence.
+  const int kRunTimeMs = 4500;
+
+  // We check that, after a `kCheckTimeMs` milliseconds (given that the CNG in
+  // Opus needs time to adapt), the absolute values of DTX decoded signal are
+  // bounded by `kOutputValueBound`.
+  const int kCheckTimeMs = 4000;
+
+#if defined(OPUS_FIXED_POINT)
+  // Fixed-point Opus generates a random (comfort) noise, which has a less
+  // predictable value bound than its floating-point Opus. This value depends on
+  // input signal, and the time window for checking the output values (between
+  // `kCheckTimeMs` and `kRunTimeMs`).
+  const uint16_t kOutputValueBound = 30;
+
+#else
+  const uint16_t kOutputValueBound = 2;
+#endif
+
+  int time = 0;
+  while (time < kRunTimeMs) {
+    // DTX mode is maintained for maximum `max_dtx_frames` frames.
+    int i = 0;
+    for (; i < max_dtx_frames; ++i) {
+      time += block_length_ms;
+      EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                    opus_encoder_, silence, opus_decoder_,
+                                    output_data_decode, &audio_type)));
+      if (dtx) {
+        if (encoded_bytes_ > 1)
+          break;
+        EXPECT_EQ(0U, encoded_bytes_)  // Send 0 byte.
+            << "Opus should have entered DTX mode.";
+        EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
+        EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
+        EXPECT_EQ(2, audio_type);  // Comfort noise.
+        if (time >= kCheckTimeMs) {
+          CheckAudioBounded(output_data_decode, output_samples, channels_,
+                            kOutputValueBound);
+        }
+      } else {
+        EXPECT_GT(encoded_bytes_, 1U);
+        EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+        EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+        EXPECT_EQ(0, audio_type);  // Speech.
+      }
+    }
+
+    if (dtx) {
+      // With DTX, Opus must stop transmission for some time.
+      EXPECT_GT(i, 1);
+    }
+
+    // We expect a normal payload.
+    EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+    EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+    EXPECT_EQ(0, audio_type);  // Speech.
+
+    // Enters DTX again immediately.
+    time += block_length_ms;
+    EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                  opus_encoder_, silence, opus_decoder_,
+                                  output_data_decode, &audio_type)));
+    if (dtx) {
+      EXPECT_EQ(1U, encoded_bytes_);  // Send 1 byte.
+      EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(2, audio_type);  // Comfort noise.
+      if (time >= kCheckTimeMs) {
+        CheckAudioBounded(output_data_decode, output_samples, channels_,
+                          kOutputValueBound);
+      }
+    } else {
+      EXPECT_GT(encoded_bytes_, 1U);
+      EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(0, audio_type);  // Speech.
+    }
+  }
+
+  silence[0] = 10000;
+  if (dtx) {
+    // Verify that encoder/decoder can jump out from DTX mode.
+    EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                  opus_encoder_, silence, opus_decoder_,
+                                  output_data_decode, &audio_type)));
+    EXPECT_GT(encoded_bytes_, 1U);
+    EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+    EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+    EXPECT_EQ(0, audio_type);  // Speech.
+  }
+
+  // Free memory.
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+// Test if CBR does what we expect.
+void OpusTest::TestCbrEffect(bool cbr, int block_length_ms) {
+  PrepareSpeechData(block_length_ms, 2000);
+  const size_t output_samples =
+      rtc::CheckedDivExact(decoder_sample_rate_hz_, 1000) * block_length_ms;
+
+  int32_t max_pkt_size_diff = 0;
+  int32_t prev_pkt_size = 0;
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Setting CBR.
+  EXPECT_EQ(0, cbr ? WebRtcOpus_EnableCbr(opus_encoder_)
+                   : WebRtcOpus_DisableCbr(opus_encoder_));
+
+  int16_t audio_type;
+  std::vector<int16_t> audio_out(output_samples * channels_);
+  for (int i = 0; i < 100; ++i) {
+    EXPECT_EQ(output_samples,
+              static_cast<size_t>(
+                  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                               opus_decoder_, audio_out.data(), &audio_type)));
+
+    if (prev_pkt_size > 0) {
+      int32_t diff = std::abs((int32_t)encoded_bytes_ - prev_pkt_size);
+      max_pkt_size_diff = std::max(max_pkt_size_diff, diff);
+    }
+    prev_pkt_size = rtc::checked_cast<int32_t>(encoded_bytes_);
+  }
+
+  if (cbr) {
+    EXPECT_EQ(max_pkt_size_diff, 0);
+  } else {
+    EXPECT_GT(max_pkt_size_diff, 0);
+  }
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+// Test failing Create.
+TEST(OpusTest, OpusCreateFail) {
+  WebRtcOpusEncInst* opus_encoder;
+  WebRtcOpusDecInst* opus_decoder;
+
+  // Test to see that an invalid pointer is caught.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(NULL, 1, 0, 48000));
+  // Invalid channel number.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 257, 0, 48000));
+  // Invalid applciation mode.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 2, 48000));
+  // Invalid sample rate.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 0, 12345));
+
+  EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(NULL, 1, 48000));
+  // Invalid channel number.
+  EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 257, 48000));
+  // Invalid sample rate.
+  EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 1, 12345));
+}
+
+// Test failing Free.
+TEST(OpusTest, OpusFreeFail) {
+  // Test to see that an invalid pointer is caught.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderFree(NULL));
+  EXPECT_EQ(-1, WebRtcOpus_DecoderFree(NULL));
+}
+
+// Test normal Create and Free.
+TEST_P(OpusTest, OpusCreateFree) {
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+  EXPECT_TRUE(opus_encoder_ != NULL);
+  EXPECT_TRUE(opus_decoder_ != NULL);
+  // Free encoder and decoder memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+#define ENCODER_CTL(inst, vargs)               \
+  inst->encoder                                \
+      ? opus_encoder_ctl(inst->encoder, vargs) \
+      : opus_multistream_encoder_ctl(inst->multistream_encoder, vargs)
+
+TEST_P(OpusTest, OpusEncodeDecode) {
+  PrepareSpeechData(20, 20);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Check number of channels for decoder.
+  EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
+
+  // Check application mode.
+  opus_int32 app;
+  ENCODER_CTL(opus_encoder_, OPUS_GET_APPLICATION(&app));
+  EXPECT_EQ(application_ == 0 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO,
+            app);
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  int16_t* output_data_decode =
+      new int16_t[decode_samples_per_channel * channels_];
+  EXPECT_EQ(decode_samples_per_channel,
+            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                         opus_decoder_, output_data_decode, &audio_type));
+
+  // Free memory.
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusSetBitRate) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
+
+  // Create encoder memory, try with different bitrates.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 30000));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 300000));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 600000));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusSetComplexity) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 9));
+
+  // Create encoder memory, try with different complexities.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 0));
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 10));
+  EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 11));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusSetBandwidth) {
+  if (channels_ > 2) {
+    // TODO(webrtc:10217): investigate why multi-stream Opus reports
+    // narrowband when it's configured with FULLBAND.
+    return;
+  }
+  PrepareSpeechData(20, 20);
+
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  std::unique_ptr<int16_t[]> output_data_decode(
+      new int16_t[decode_samples_per_channel * channels_]());
+
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1,
+            WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND));
+  EXPECT_EQ(-1, WebRtcOpus_GetBandwidth(opus_encoder_));
+
+  // Create encoder memory, try with different bandwidths.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  EXPECT_EQ(-1, WebRtcOpus_SetBandwidth(opus_encoder_,
+                                        OPUS_BANDWIDTH_NARROWBAND - 1));
+  EXPECT_EQ(0,
+            WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND));
+  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+               output_data_decode.get(), &audio_type);
+  EXPECT_EQ(OPUS_BANDWIDTH_NARROWBAND, WebRtcOpus_GetBandwidth(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_FULLBAND));
+  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+               output_data_decode.get(), &audio_type);
+  EXPECT_EQ(encoder_sample_rate_hz_ == 16000 ? OPUS_BANDWIDTH_WIDEBAND
+                                             : OPUS_BANDWIDTH_FULLBAND,
+            WebRtcOpus_GetBandwidth(opus_encoder_));
+  EXPECT_EQ(
+      -1, WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_FULLBAND + 1));
+  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+               output_data_decode.get(), &audio_type);
+  EXPECT_EQ(encoder_sample_rate_hz_ == 16000 ? OPUS_BANDWIDTH_WIDEBAND
+                                             : OPUS_BANDWIDTH_FULLBAND,
+            WebRtcOpus_GetBandwidth(opus_encoder_));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusForceChannels) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
+
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  ASSERT_NE(nullptr, opus_encoder_);
+
+  if (channels_ >= 2) {
+    EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 3));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 2));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 0));
+  } else {
+    EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 2));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 0));
+  }
+
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+// Encode and decode one frame, initialize the decoder and
+// decode once more.
+TEST_P(OpusTest, OpusDecodeInit) {
+  PrepareSpeechData(20, 20);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  int16_t* output_data_decode =
+      new int16_t[decode_samples_per_channel * channels_];
+  EXPECT_EQ(decode_samples_per_channel,
+            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                         opus_decoder_, output_data_decode, &audio_type));
+
+  WebRtcOpus_DecoderInit(opus_decoder_);
+
+  EXPECT_EQ(decode_samples_per_channel,
+            WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
+                              output_data_decode, &audio_type));
+
+  // Free memory.
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusEnableDisableFec) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_EnableFec(opus_encoder_));
+  EXPECT_EQ(-1, WebRtcOpus_DisableFec(opus_encoder_));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusEnableDisableDtx) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_EnableDtx(opus_encoder_));
+  EXPECT_EQ(-1, WebRtcOpus_DisableDtx(opus_encoder_));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  opus_int32 dtx;
+
+  // DTX is off by default.
+  ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
+  EXPECT_EQ(0, dtx);
+
+  // Test to enable DTX.
+  EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
+  ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
+  EXPECT_EQ(1, dtx);
+
+  // Test to disable DTX.
+  EXPECT_EQ(0, WebRtcOpus_DisableDtx(opus_encoder_));
+  ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
+  EXPECT_EQ(0, dtx);
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusDtxOff) {
+  TestDtxEffect(false, 10);
+  TestDtxEffect(false, 20);
+  TestDtxEffect(false, 40);
+}
+
+TEST_P(OpusTest, OpusDtxOn) {
+  if (channels_ > 2 || application_ != 0) {
+    // DTX does not work with OPUS_APPLICATION_AUDIO at low complexity settings.
+    // TODO(webrtc:10218): adapt the test to the sizes and order of multi-stream
+    // DTX packets.
+    return;
+  }
+  TestDtxEffect(true, 10);
+  TestDtxEffect(true, 20);
+  TestDtxEffect(true, 40);
+}
+
+TEST_P(OpusTest, OpusCbrOff) {
+  TestCbrEffect(false, 10);
+  TestCbrEffect(false, 20);
+  TestCbrEffect(false, 40);
+}
+
+TEST_P(OpusTest, OpusCbrOn) {
+  TestCbrEffect(true, 10);
+  TestCbrEffect(true, 20);
+  TestCbrEffect(true, 40);
+}
+
+TEST_P(OpusTest, OpusSetPacketLossRate) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
+  EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, -1));
+  EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 101));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusSetMaxPlaybackRate) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, 20000));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 48000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 24001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_SUPERWIDEBAND, 24000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_SUPERWIDEBAND, 16001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_WIDEBAND, 16000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_WIDEBAND, 12001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_MEDIUMBAND, 12000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_MEDIUMBAND, 8001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND, 8000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND, 4000);
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+// Test PLC.
+TEST_P(OpusTest, OpusDecodePlc) {
+  PrepareSpeechData(20, 20);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Check number of channels for decoder.
+  EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  int16_t* output_data_decode =
+      new int16_t[decode_samples_per_channel * channels_];
+  EXPECT_EQ(decode_samples_per_channel,
+            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                         opus_decoder_, output_data_decode, &audio_type));
+
+  // Call decoder PLC.
+  constexpr int kPlcDurationMs = 10;
+  const int plc_samples = decoder_sample_rate_hz_ * kPlcDurationMs / 1000;
+  int16_t* plc_buffer = new int16_t[plc_samples * channels_];
+  EXPECT_EQ(plc_samples,
+            WebRtcOpus_Decode(opus_decoder_, NULL, 0, plc_buffer, &audio_type));
+
+  // Free memory.
+  delete[] plc_buffer;
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+// Duration estimation.
+TEST_P(OpusTest, OpusDurationEstimation) {
+  PrepareSpeechData(20, 20);
+
+  // Create.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // 10 ms. We use only first 10 ms of a 20 ms block.
+  auto speech_block = speech_data_.GetNextBlock();
+  int encoded_bytes_int = WebRtcOpus_Encode(
+      opus_encoder_, speech_block.data(),
+      rtc::CheckedDivExact(speech_block.size(), 2 * channels_), kMaxBytes,
+      bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
+  EXPECT_EQ(SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/10),
+            WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
+                                   static_cast<size_t>(encoded_bytes_int)));
+
+  // 20 ms
+  speech_block = speech_data_.GetNextBlock();
+  encoded_bytes_int =
+      WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+                        rtc::CheckedDivExact(speech_block.size(), channels_),
+                        kMaxBytes, bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
+  EXPECT_EQ(SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20),
+            WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
+                                   static_cast<size_t>(encoded_bytes_int)));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusDecodeRepacketized) {
+  if (channels_ > 2) {
+    // As per the Opus documentation
+    // https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__repacketizer.html#details,
+    // multiple streams are not supported.
+    return;
+  }
+  constexpr size_t kPackets = 6;
+
+  PrepareSpeechData(20, 20 * kPackets);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  ASSERT_NE(nullptr, opus_encoder_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+  ASSERT_NE(nullptr, opus_decoder_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Check number of channels for decoder.
+  EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  std::unique_ptr<int16_t[]> output_data_decode(
+      new int16_t[kPackets * decode_samples_per_channel * channels_]);
+  OpusRepacketizer* rp = opus_repacketizer_create();
+
+  size_t num_packets = 0;
+  constexpr size_t kMaxCycles = 100;
+  for (size_t idx = 0; idx < kMaxCycles; ++idx) {
+    auto speech_block = speech_data_.GetNextBlock();
+    encoded_bytes_ =
+        WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+                          rtc::CheckedDivExact(speech_block.size(), channels_),
+                          kMaxBytes, bitstream_);
+    if (opus_repacketizer_cat(rp, bitstream_,
+                              rtc::checked_cast<opus_int32>(encoded_bytes_)) ==
+        OPUS_OK) {
+      ++num_packets;
+      if (num_packets == kPackets) {
+        break;
+      }
+    } else {
+      // Opus repacketizer cannot guarantee a success. We try again if it fails.
+      opus_repacketizer_init(rp);
+      num_packets = 0;
+    }
+  }
+  EXPECT_EQ(kPackets, num_packets);
+
+  encoded_bytes_ = opus_repacketizer_out(rp, bitstream_, kMaxBytes);
+
+  EXPECT_EQ(decode_samples_per_channel * kPackets,
+            static_cast<size_t>(WebRtcOpus_DurationEst(
+                opus_decoder_, bitstream_, encoded_bytes_)));
+
+  EXPECT_EQ(decode_samples_per_channel * kPackets,
+            static_cast<size_t>(
+                WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
+                                  output_data_decode.get(), &audio_type)));
+
+  // Free memory.
+  opus_repacketizer_destroy(rp);
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST(OpusVadTest, CeltUnknownStatus) {
+  const uint8_t celt[] = {0x80};
+  EXPECT_EQ(WebRtcOpus_PacketHasVoiceActivity(celt, 1), -1);
+}
+
+TEST(OpusVadTest, Mono20msVadSet) {
+  uint8_t silk20msMonoVad[] = {0x78, 0x80};
+  EXPECT_TRUE(WebRtcOpus_PacketHasVoiceActivity(silk20msMonoVad, 2));
+}
+
+TEST(OpusVadTest, Mono20MsVadUnset) {
+  uint8_t silk20msMonoSilence[] = {0x78, 0x00};
+  EXPECT_FALSE(WebRtcOpus_PacketHasVoiceActivity(silk20msMonoSilence, 2));
+}
+
+TEST(OpusVadTest, Stereo20MsVadOnSideChannel) {
+  uint8_t silk20msStereoVadSideChannel[] = {0x78 | 0x04, 0x20};
+  EXPECT_TRUE(
+      WebRtcOpus_PacketHasVoiceActivity(silk20msStereoVadSideChannel, 2));
+}
+
+TEST(OpusVadTest, TwoOpusMonoFramesVadOnSecond) {
+  uint8_t twoMonoFrames[] = {0x78 | 0x1, 0x00, 0x80};
+  EXPECT_TRUE(WebRtcOpus_PacketHasVoiceActivity(twoMonoFrames, 3));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn
new file mode 100644
index 0000000000..8bc0bf5e0e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn
@@ -0,0 +1,55 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../../../webrtc.gni")
+
+visibility = [
+  ":*",
+  "../../../:*",
+]
+
+if (rtc_include_tests) {
+  rtc_library("test") {
+    testonly = true
+
+    sources = [
+      "audio_ring_buffer.cc",
+      "audio_ring_buffer.h",
+      "blocker.cc",
+      "blocker.h",
+      "lapped_transform.cc",
+      "lapped_transform.h",
+    ]
+
+    deps = [
+      "../../../../../common_audio",
+      "../../../../../common_audio:common_audio_c",
+      "../../../../../rtc_base:checks",
+      "../../../../../rtc_base/memory:aligned_malloc",
+    ]
+  }
+
+  rtc_library("test_unittest") {
+    testonly = true
+
+    sources = [
+      "audio_ring_buffer_unittest.cc",
+      "blocker_unittest.cc",
+      "lapped_transform_unittest.cc",
+    ]
+
+    deps = [
+      ":test",
+      "../../../../../common_audio",
+      "../../../../../common_audio:common_audio_c",
+      "../../../../../rtc_base:macromagic",
+      "../../../../../test:test_support",
+      "//testing/gtest",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc
new file mode 100644
index 0000000000..2a71b43d2c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+
+#include "common_audio/ring_buffer.h"
+#include "rtc_base/checks.h"
+
+// This is a simple multi-channel wrapper over the ring_buffer.h C interface.
+
+namespace webrtc {
+
+AudioRingBuffer::AudioRingBuffer(size_t channels, size_t max_frames) {
+  buffers_.reserve(channels);
+  for (size_t i = 0; i < channels; ++i)
+    buffers_.push_back(WebRtc_CreateBuffer(max_frames, sizeof(float)));
+}
+
+AudioRingBuffer::~AudioRingBuffer() {
+  for (auto* buf : buffers_)
+    WebRtc_FreeBuffer(buf);
+}
+
+void AudioRingBuffer::Write(const float* const* data,
+                            size_t channels,
+                            size_t frames) {
+  RTC_DCHECK_EQ(buffers_.size(), channels);
+  for (size_t i = 0; i < channels; ++i) {
+    const size_t written = WebRtc_WriteBuffer(buffers_[i], data[i], frames);
+    RTC_CHECK_EQ(written, frames);
+  }
+}
+
+void AudioRingBuffer::Read(float* const* data, size_t channels, size_t frames) {
+  RTC_DCHECK_EQ(buffers_.size(), channels);
+  for (size_t i = 0; i < channels; ++i) {
+    const size_t read =
+        WebRtc_ReadBuffer(buffers_[i], nullptr, data[i], frames);
+    RTC_CHECK_EQ(read, frames);
+  }
+}
+
+size_t AudioRingBuffer::ReadFramesAvailable() const {
+  // All buffers have the same amount available.
+  return WebRtc_available_read(buffers_[0]);
+}
+
+size_t AudioRingBuffer::WriteFramesAvailable() const {
+  // All buffers have the same amount available.
+  return WebRtc_available_write(buffers_[0]);
+}
+
+void AudioRingBuffer::MoveReadPositionForward(size_t frames) {
+  for (auto* buf : buffers_) {
+    const size_t moved =
+        static_cast<size_t>(WebRtc_MoveReadPtr(buf, static_cast<int>(frames)));
+    RTC_CHECK_EQ(moved, frames);
+  }
+}
+
+void AudioRingBuffer::MoveReadPositionBackward(size_t frames) {
+  for (auto* buf : buffers_) {
+    const size_t moved = static_cast<size_t>(
+        -WebRtc_MoveReadPtr(buf, -static_cast<int>(frames)));
+    RTC_CHECK_EQ(moved, frames);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h
new file mode 100644
index 0000000000..a280ca2410
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+struct RingBuffer;
+
+namespace webrtc {
+
+// A ring buffer tailored for float deinterleaved audio. Any operation that
+// cannot be performed as requested will cause a crash (e.g. insufficient data
+// in the buffer to fulfill a read request.)
+class AudioRingBuffer final {
+ public:
+  // Specify the number of channels and maximum number of frames the buffer will
+  // contain.
+  AudioRingBuffer(size_t channels, size_t max_frames);
+  ~AudioRingBuffer();
+
+  // Copies `data` to the buffer and advances the write pointer. `channels` must
+  // be the same as at creation time.
+  void Write(const float* const* data, size_t channels, size_t frames);
+
+  // Copies from the buffer to `data` and advances the read pointer. `channels`
+  // must be the same as at creation time.
+  void Read(float* const* data, size_t channels, size_t frames);
+
+  size_t ReadFramesAvailable() const;
+  size_t WriteFramesAvailable() const;
+
+  // Moves the read position. The forward version advances the read pointer
+  // towards the write pointer and the backward verison withdraws the read
+  // pointer away from the write pointer (i.e. flushing and stuffing the buffer
+  // respectively.)
+  void MoveReadPositionForward(size_t frames);
+  void MoveReadPositionBackward(size_t frames);
+
+ private:
+  // TODO(kwiberg): Use std::vector<std::unique_ptr<RingBuffer>> instead.
+  std::vector<RingBuffer*> buffers_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
new file mode 100644
index 0000000000..6dbc8ee9fe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+
+#include <memory>
+
+#include "common_audio/channel_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class AudioRingBufferTest
+    : public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > {
+};
+
+void ReadAndWriteTest(const ChannelBuffer<float>& input,
+                      size_t num_write_chunk_frames,
+                      size_t num_read_chunk_frames,
+                      size_t buffer_frames,
+                      ChannelBuffer<float>* output) {
+  const size_t num_channels = input.num_channels();
+  const size_t total_frames = input.num_frames();
+  AudioRingBuffer buf(num_channels, buffer_frames);
+  std::unique_ptr<float*[]> slice(new float*[num_channels]);
+
+  size_t input_pos = 0;
+  size_t output_pos = 0;
+  while (input_pos + buf.WriteFramesAvailable() < total_frames) {
+    // Write until the buffer is as full as possible.
+    while (buf.WriteFramesAvailable() >= num_write_chunk_frames) {
+      buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+                num_write_chunk_frames);
+      input_pos += num_write_chunk_frames;
+    }
+    // Read until the buffer is as empty as possible.
+    while (buf.ReadFramesAvailable() >= num_read_chunk_frames) {
+      EXPECT_LT(output_pos, total_frames);
+      buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+               num_read_chunk_frames);
+      output_pos += num_read_chunk_frames;
+    }
+  }
+
+  // Write and read the last bit.
+  if (input_pos < total_frames) {
+    buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+              total_frames - input_pos);
+  }
+  if (buf.ReadFramesAvailable()) {
+    buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+             buf.ReadFramesAvailable());
+  }
+  EXPECT_EQ(0u, buf.ReadFramesAvailable());
+}
+
+TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
+  const size_t kFrames = 5000;
+  const size_t num_channels = ::testing::get<3>(GetParam());
+
+  // Initialize the input data to an increasing sequence.
+  ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
+  for (size_t i = 0; i < num_channels; ++i)
+    for (size_t j = 0; j < kFrames; ++j)
+      input.channels()[i][j] = (i + 1) * (j + 1);
+
+  ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels));
+  ReadAndWriteTest(input, ::testing::get<0>(GetParam()),
+                   ::testing::get<1>(GetParam()), ::testing::get<2>(GetParam()),
+                   &output);
+
+  // Verify the read data matches the input.
+  for (size_t i = 0; i < num_channels; ++i)
+    for (size_t j = 0; j < kFrames; ++j)
+      EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioRingBufferTest,
+    AudioRingBufferTest,
+    ::testing::Combine(::testing::Values(10, 20, 42),  // num_write_chunk_frames
+                       ::testing::Values(1, 10, 17),   // num_read_chunk_frames
+                       ::testing::Values(100, 256),    // buffer_frames
+                       ::testing::Values(1, 4)));      // num_channels
+
+TEST_F(AudioRingBufferTest, MoveReadPosition) {
+  const size_t kNumChannels = 1;
+  const float kInputArray[] = {1, 2, 3, 4};
+  const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray);
+  ChannelBuffer<float> input(kNumFrames, kNumChannels);
+  input.SetDataForTesting(kInputArray, kNumFrames);
+  AudioRingBuffer buf(kNumChannels, kNumFrames);
+  buf.Write(input.channels(), kNumChannels, kNumFrames);
+
+  buf.MoveReadPositionForward(3);
+  ChannelBuffer<float> output(1, kNumChannels);
+  buf.Read(output.channels(), kNumChannels, 1);
+  EXPECT_EQ(4, output.channels()[0][0]);
+  buf.MoveReadPositionBackward(3);
+  buf.Read(output.channels(), kNumChannels, 1);
+  EXPECT_EQ(2, output.channels()[0][0]);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc
new file mode 100644
index 0000000000..33406cead9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc
@@ -0,0 +1,215 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/blocker.h"
+
+#include <string.h>
+
+#include "rtc_base/checks.h"
+
+namespace {
+
+// Adds `a` and `b` frame by frame into `result` (basically matrix addition).
+void AddFrames(const float* const* a,
+               size_t a_start_index,
+               const float* const* b,
+               int b_start_index,
+               size_t num_frames,
+               size_t num_channels,
+               float* const* result,
+               size_t result_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    for (size_t j = 0; j < num_frames; ++j) {
+      result[i][j + result_start_index] =
+          a[i][j + a_start_index] + b[i][j + b_start_index];
+    }
+  }
+}
+
+// Copies `src` into `dst` channel by channel.
+void CopyFrames(const float* const* src,
+                size_t src_start_index,
+                size_t num_frames,
+                size_t num_channels,
+                float* const* dst,
+                size_t dst_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    memcpy(&dst[i][dst_start_index], &src[i][src_start_index],
+           num_frames * sizeof(dst[i][dst_start_index]));
+  }
+}
+
+// Moves `src` into `dst` channel by channel.
+void MoveFrames(const float* const* src,
+                size_t src_start_index,
+                size_t num_frames,
+                size_t num_channels,
+                float* const* dst,
+                size_t dst_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    memmove(&dst[i][dst_start_index], &src[i][src_start_index],
+            num_frames * sizeof(dst[i][dst_start_index]));
+  }
+}
+
+void ZeroOut(float* const* buffer,
+             size_t starting_idx,
+             size_t num_frames,
+             size_t num_channels) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    memset(&buffer[i][starting_idx], 0,
+           num_frames * sizeof(buffer[i][starting_idx]));
+  }
+}
+
+// Pointwise multiplies each channel of `frames` with `window`. Results are
+// stored in `frames`.
+void ApplyWindow(const float* window,
+                 size_t num_frames,
+                 size_t num_channels,
+                 float* const* frames) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    for (size_t j = 0; j < num_frames; ++j) {
+      frames[i][j] = frames[i][j] * window[j];
+    }
+  }
+}
+
+size_t gcd(size_t a, size_t b) {
+  size_t tmp;
+  while (b) {
+    tmp = a;
+    a = b;
+    b = tmp % b;
+  }
+  return a;
+}
+
+}  // namespace
+
+namespace webrtc {
+
+Blocker::Blocker(size_t chunk_size,
+                 size_t block_size,
+                 size_t num_input_channels,
+                 size_t num_output_channels,
+                 const float* window,
+                 size_t shift_amount,
+                 BlockerCallback* callback)
+    : chunk_size_(chunk_size),
+      block_size_(block_size),
+      num_input_channels_(num_input_channels),
+      num_output_channels_(num_output_channels),
+      initial_delay_(block_size_ - gcd(chunk_size, shift_amount)),
+      frame_offset_(0),
+      input_buffer_(num_input_channels_, chunk_size_ + initial_delay_),
+      output_buffer_(chunk_size_ + initial_delay_, num_output_channels_),
+      input_block_(block_size_, num_input_channels_),
+      output_block_(block_size_, num_output_channels_),
+      window_(new float[block_size_]),
+      shift_amount_(shift_amount),
+      callback_(callback) {
+  RTC_CHECK_LE(num_output_channels_, num_input_channels_);
+  RTC_CHECK_LE(shift_amount_, block_size_);
+
+  memcpy(window_.get(), window, block_size_ * sizeof(*window_.get()));
+  input_buffer_.MoveReadPositionBackward(initial_delay_);
+}
+
+Blocker::~Blocker() = default;
+
+// When block_size < chunk_size the input and output buffers look like this:
+//
+//                      delay*             chunk_size    chunk_size + delay*
+//  buffer: <-------------|---------------------|---------------|>
+//                _a_              _b_                 _c_
+//
+// On each call to ProcessChunk():
+// 1. New input gets read into sections _b_ and _c_ of the input buffer.
+// 2. We block starting from frame_offset.
+// 3. We block until we reach a block `bl` that doesn't contain any frames
+//    from sections _a_ or _b_ of the input buffer.
+// 4. We window the current block, fire the callback for processing, window
+//    again, and overlap/add to the output buffer.
+// 5. We copy sections _a_ and _b_ of the output buffer into output.
+// 6. For both the input and the output buffers, we copy section _c_ into
+//    section _a_.
+// 7. We set the new frame_offset to be the difference between the first frame
+//    of `bl` and the border between sections _b_ and _c_.
+//
+// When block_size > chunk_size the input and output buffers look like this:
+//
+//                   chunk_size               delay*       chunk_size + delay*
+//  buffer: <-------------|---------------------|---------------|>
+//                _a_              _b_                 _c_
+//
+// On each call to ProcessChunk():
+// The procedure is the same as above, except for:
+// 1. New input gets read into section _c_ of the input buffer.
+// 3. We block until we reach a block `bl` that doesn't contain any frames
+//    from section _a_ of the input buffer.
+// 5. We copy section _a_ of the output buffer into output.
+// 6. For both the input and the output buffers, we copy sections _b_ and _c_
+//    into section _a_ and _b_.
+// 7. We set the new frame_offset to be the difference between the first frame
+//    of `bl` and the border between sections _a_ and _b_.
+//
+// * delay here refers to inintial_delay_
+//
+// TODO(claguna): Look at using ring buffers to eliminate some copies.
+void Blocker::ProcessChunk(const float* const* input,
+                           size_t chunk_size,
+                           size_t num_input_channels,
+                           size_t num_output_channels,
+                           float* const* output) {
+  RTC_CHECK_EQ(chunk_size, chunk_size_);
+  RTC_CHECK_EQ(num_input_channels, num_input_channels_);
+  RTC_CHECK_EQ(num_output_channels, num_output_channels_);
+
+  input_buffer_.Write(input, num_input_channels, chunk_size_);
+  size_t first_frame_in_block = frame_offset_;
+
+  // Loop through blocks.
+  while (first_frame_in_block < chunk_size_) {
+    input_buffer_.Read(input_block_.channels(), num_input_channels,
+                       block_size_);
+    input_buffer_.MoveReadPositionBackward(block_size_ - shift_amount_);
+
+    ApplyWindow(window_.get(), block_size_, num_input_channels_,
+                input_block_.channels());
+    callback_->ProcessBlock(input_block_.channels(), block_size_,
+                            num_input_channels_, num_output_channels_,
+                            output_block_.channels());
+    ApplyWindow(window_.get(), block_size_, num_output_channels_,
+                output_block_.channels());
+
+    AddFrames(output_buffer_.channels(), first_frame_in_block,
+              output_block_.channels(), 0, block_size_, num_output_channels_,
+              output_buffer_.channels(), first_frame_in_block);
+
+    first_frame_in_block += shift_amount_;
+  }
+
+  // Copy output buffer to output
+  CopyFrames(output_buffer_.channels(), 0, chunk_size_, num_output_channels_,
+             output, 0);
+
+  // Copy output buffer [chunk_size_, chunk_size_ + initial_delay]
+  // to output buffer [0, initial_delay], zero the rest.
+  MoveFrames(output_buffer_.channels(), chunk_size, initial_delay_,
+             num_output_channels_, output_buffer_.channels(), 0);
+  ZeroOut(output_buffer_.channels(), initial_delay_, chunk_size_,
+          num_output_channels_);
+
+  // Calculate new starting frames.
+  frame_offset_ = first_frame_in_block - chunk_size_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h
new file mode 100644
index 0000000000..59b7e29621
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
+
+#include <memory>
+
+#include "common_audio/channel_buffer.h"
+#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+
+namespace webrtc {
+
+// The callback function to process audio in the time domain. Input has already
+// been windowed, and output will be windowed. The number of input channels
+// must be >= the number of output channels.
+class BlockerCallback {
+ public:
+  virtual ~BlockerCallback() {}
+
+  virtual void ProcessBlock(const float* const* input,
+                            size_t num_frames,
+                            size_t num_input_channels,
+                            size_t num_output_channels,
+                            float* const* output) = 0;
+};
+
+// The main purpose of Blocker is to abstract away the fact that often we
+// receive a different number of audio frames than our transform takes. For
+// example, most FFTs work best when the fft-size is a power of 2, but suppose
+// we receive 20ms of audio at a sample rate of 48000. That comes to 960 frames
+// of audio, which is not a power of 2. Blocker allows us to specify the
+// transform and all other necessary processing via the Process() callback
+// function without any constraints on the transform-size
+// (read: `block_size_`) or received-audio-size (read: `chunk_size_`).
+// We handle this for the multichannel audio case, allowing for different
+// numbers of input and output channels (for example, beamforming takes 2 or
+// more input channels and returns 1 output channel). Audio signals are
+// represented as deinterleaved floats in the range [-1, 1].
+//
+// Blocker is responsible for:
+// - blocking audio while handling potential discontinuities on the edges
+//   of chunks
+// - windowing blocks before sending them to Process()
+// - windowing processed blocks, and overlap-adding them together before
+//   sending back a processed chunk
+//
+// To use blocker:
+// 1. Impelment a BlockerCallback object `bc`.
+// 2. Instantiate a Blocker object `b`, passing in `bc`.
+// 3. As you receive audio, call b.ProcessChunk() to get processed audio.
+//
+// A small amount of delay is added to the first received chunk to deal with
+// the difference in chunk/block sizes. This delay is <= chunk_size.
+//
+// Ownership of window is retained by the caller.  That is, Blocker makes a
+// copy of window and does not attempt to delete it.
+class Blocker {
+ public:
+  Blocker(size_t chunk_size,
+          size_t block_size,
+          size_t num_input_channels,
+          size_t num_output_channels,
+          const float* window,
+          size_t shift_amount,
+          BlockerCallback* callback);
+  ~Blocker();
+
+  void ProcessChunk(const float* const* input,
+                    size_t chunk_size,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
+                    float* const* output);
+
+  size_t initial_delay() const { return initial_delay_; }
+
+ private:
+  const size_t chunk_size_;
+  const size_t block_size_;
+  const size_t num_input_channels_;
+  const size_t num_output_channels_;
+
+  // The number of frames of delay to add at the beginning of the first chunk.
+  const size_t initial_delay_;
+
+  // The frame index into the input buffer where the first block should be read
+  // from. This is necessary because shift_amount_ is not necessarily a
+  // multiple of chunk_size_, so blocks won't line up at the start of the
+  // buffer.
+  size_t frame_offset_;
+
+  // Since blocks nearly always overlap, there are certain blocks that require
+  // frames from the end of one chunk and the beginning of the next chunk. The
+  // input and output buffers are responsible for saving those frames between
+  // calls to ProcessChunk().
+  //
+  // Both contain |initial delay| + `chunk_size` frames. The input is a fairly
+  // standard FIFO, but due to the overlap-add it's harder to use an
+  // AudioRingBuffer for the output.
+  AudioRingBuffer input_buffer_;
+  ChannelBuffer<float> output_buffer_;
+
+  // Space for the input block (can't wrap because of windowing).
+  ChannelBuffer<float> input_block_;
+
+  // Space for the output block (can't wrap because of overlap/add).
+  ChannelBuffer<float> output_block_;
+
+  std::unique_ptr<float[]> window_;
+
+  // The amount of frames between the start of contiguous blocks. For example,
+  // `shift_amount_` = `block_size_` / 2 for a Hann window.
+  size_t shift_amount_;
+
+  BlockerCallback* callback_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
new file mode 100644
index 0000000000..9c8e789ba9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
@@ -0,0 +1,293 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/blocker.h"
+
+#include <memory>
+
+#include "rtc_base/arraysize.h"
+#include "test/gtest.h"
+
+namespace {
+
+// Callback Function to add 3 to every sample in the signal.
+class PlusThreeBlockerCallback : public webrtc::BlockerCallback {
+ public:
+  void ProcessBlock(const float* const* input,
+                    size_t num_frames,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
+                    float* const* output) override {
+    for (size_t i = 0; i < num_output_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        output[i][j] = input[i][j] + 3;
+      }
+    }
+  }
+};
+
+// No-op Callback Function.
+class CopyBlockerCallback : public webrtc::BlockerCallback {
+ public:
+  void ProcessBlock(const float* const* input,
+                    size_t num_frames,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
+                    float* const* output) override {
+    for (size_t i = 0; i < num_output_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        output[i][j] = input[i][j];
+      }
+    }
+  }
+};
+
+}  // namespace
+
+namespace webrtc {
+
+// Tests blocking with a window that multiplies the signal by 2, a callback
+// that adds 3 to each sample in the signal, and different combinations of chunk
+// size, block size, and shift amount.
+class BlockerTest : public ::testing::Test {
+ protected:
+  void RunTest(Blocker* blocker,
+               size_t chunk_size,
+               size_t num_frames,
+               const float* const* input,
+               float* const* input_chunk,
+               float* const* output,
+               float* const* output_chunk,
+               size_t num_input_channels,
+               size_t num_output_channels) {
+    size_t start = 0;
+    size_t end = chunk_size - 1;
+    while (end < num_frames) {
+      CopyTo(input_chunk, 0, start, num_input_channels, chunk_size, input);
+      blocker->ProcessChunk(input_chunk, chunk_size, num_input_channels,
+                            num_output_channels, output_chunk);
+      CopyTo(output, start, 0, num_output_channels, chunk_size, output_chunk);
+
+      start += chunk_size;
+      end += chunk_size;
+    }
+  }
+
+  void ValidateSignalEquality(const float* const* expected,
+                              const float* const* actual,
+                              size_t num_channels,
+                              size_t num_frames) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        EXPECT_FLOAT_EQ(expected[i][j], actual[i][j]);
+      }
+    }
+  }
+
+  void ValidateInitialDelay(const float* const* output,
+                            size_t num_channels,
+                            size_t num_frames,
+                            size_t initial_delay) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        if (j < initial_delay) {
+          EXPECT_FLOAT_EQ(output[i][j], 0.f);
+        } else {
+          EXPECT_GT(output[i][j], 0.f);
+        }
+      }
+    }
+  }
+
+  static void CopyTo(float* const* dst,
+                     size_t start_index_dst,
+                     size_t start_index_src,
+                     size_t num_channels,
+                     size_t num_frames,
+                     const float* const* src) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      memcpy(&dst[i][start_index_dst], &src[i][start_index_src],
+             num_frames * sizeof(float));
+    }
+  }
+};
+
+TEST_F(BlockerTest, TestBlockerMutuallyPrimeChunkandBlockSize) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 10;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 5;
+  const size_t kShiftAmount = 2;
+
+  const float kInput[kNumInputChannels][kNumFrames] = {
+      {1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
+      {2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
+      {3, 3, 3, 3, 3, 3, 3, 3, 3, 3}};
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput));
+
+  const float kExpectedOutput[kNumInputChannels][kNumFrames] = {
+      {6, 6, 12, 20, 20, 20, 20, 20, 20, 20},
+      {6, 6, 12, 28, 28, 28, 28, 28, 28, 28}};
+  ChannelBuffer<float> expected_output_cb(kNumFrames, kNumInputChannels);
+  expected_output_cb.SetDataForTesting(
+      kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput));
+
+  const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f};
+
+  ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels);
+  ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels);
+  ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
+
+  PlusThreeBlockerCallback callback;
+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
+                  kWindow, kShiftAmount, &callback);
+
+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
+          input_chunk_cb.channels(), actual_output_cb.channels(),
+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+  ValidateSignalEquality(expected_output_cb.channels(),
+                         actual_output_cb.channels(), kNumOutputChannels,
+                         kNumFrames);
+}
+
+TEST_F(BlockerTest, TestBlockerMutuallyPrimeShiftAndBlockSize) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 12;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 6;
+  const size_t kShiftAmount = 3;
+
+  const float kInput[kNumInputChannels][kNumFrames] = {
+      {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
+      {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
+      {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}};
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput));
+
+  const float kExpectedOutput[kNumOutputChannels][kNumFrames] = {
+      {6, 10, 10, 20, 10, 10, 20, 10, 10, 20, 10, 10},
+      {6, 14, 14, 28, 14, 14, 28, 14, 14, 28, 14, 14}};
+  ChannelBuffer<float> expected_output_cb(kNumFrames, kNumOutputChannels);
+  expected_output_cb.SetDataForTesting(
+      kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput));
+
+  const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f};
+
+  ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels);
+  ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels);
+  ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
+
+  PlusThreeBlockerCallback callback;
+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
+                  kWindow, kShiftAmount, &callback);
+
+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
+          input_chunk_cb.channels(), actual_output_cb.channels(),
+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+  ValidateSignalEquality(expected_output_cb.channels(),
+                         actual_output_cb.channels(), kNumOutputChannels,
+                         kNumFrames);
+}
+
+TEST_F(BlockerTest, TestBlockerNoOverlap) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 12;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 4;
+  const size_t kShiftAmount = 4;
+
+  const float kInput[kNumInputChannels][kNumFrames] = {
+      {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
+      {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
+      {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}};
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput));
+
+  const float kExpectedOutput[kNumOutputChannels][kNumFrames] = {
+      {10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10},
+      {14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14}};
+  ChannelBuffer<float> expected_output_cb(kNumFrames, kNumOutputChannels);
+  expected_output_cb.SetDataForTesting(
+      kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput));
+
+  const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f};
+
+  ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels);
+  ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels);
+  ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
+
+  PlusThreeBlockerCallback callback;
+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
+                  kWindow, kShiftAmount, &callback);
+
+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
+          input_chunk_cb.channels(), actual_output_cb.channels(),
+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+  ValidateSignalEquality(expected_output_cb.channels(),
+                         actual_output_cb.channels(), kNumOutputChannels,
+                         kNumFrames);
+}
+
+TEST_F(BlockerTest, InitialDelaysAreMinimum) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 1280;
+  const size_t kChunkSize[] = {80,  80,  80,  80,  80,  80,
+                               160, 160, 160, 160, 160, 160};
+  const size_t kBlockSize[] = {64,  64,  64,  128, 128, 128,
+                               128, 128, 128, 256, 256, 256};
+  const size_t kShiftAmount[] = {16, 32, 64,  32, 64,  128,
+                                 32, 64, 128, 64, 128, 256};
+  const size_t kInitialDelay[] = {48, 48, 48, 112, 112, 112,
+                                  96, 96, 96, 224, 224, 224};
+
+  float input[kNumInputChannels][kNumFrames];
+  for (size_t i = 0; i < kNumInputChannels; ++i) {
+    for (size_t j = 0; j < kNumFrames; ++j) {
+      input[i][j] = i + 1;
+    }
+  }
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(input[0], sizeof(input) / sizeof(**input));
+
+  ChannelBuffer<float> output_cb(kNumFrames, kNumOutputChannels);
+
+  CopyBlockerCallback callback;
+
+  for (size_t i = 0; i < arraysize(kChunkSize); ++i) {
+    std::unique_ptr<float[]> window(new float[kBlockSize[i]]);
+    for (size_t j = 0; j < kBlockSize[i]; ++j) {
+      window[j] = 1.f;
+    }
+
+    ChannelBuffer<float> input_chunk_cb(kChunkSize[i], kNumInputChannels);
+    ChannelBuffer<float> output_chunk_cb(kChunkSize[i], kNumOutputChannels);
+
+    Blocker blocker(kChunkSize[i], kBlockSize[i], kNumInputChannels,
+                    kNumOutputChannels, window.get(), kShiftAmount[i],
+                    &callback);
+
+    RunTest(&blocker, kChunkSize[i], kNumFrames, input_cb.channels(),
+            input_chunk_cb.channels(), output_cb.channels(),
+            output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+    ValidateInitialDelay(output_cb.channels(), kNumOutputChannels, kNumFrames,
+                         kInitialDelay[i]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc
new file mode 100644
index 0000000000..b1a6526bba
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
+
+#include <algorithm>
+#include <cstdlib>
+#include <cstring>
+
+#include "common_audio/real_fourier.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void LappedTransform::BlockThunk::ProcessBlock(const float* const* input,
+                                               size_t num_frames,
+                                               size_t num_input_channels,
+                                               size_t num_output_channels,
+                                               float* const* output) {
+  RTC_CHECK_EQ(num_input_channels, parent_->num_in_channels_);
+  RTC_CHECK_EQ(num_output_channels, parent_->num_out_channels_);
+  RTC_CHECK_EQ(parent_->block_length_, num_frames);
+
+  for (size_t i = 0; i < num_input_channels; ++i) {
+    memcpy(parent_->real_buf_.Row(i), input[i], num_frames * sizeof(*input[0]));
+    parent_->fft_->Forward(parent_->real_buf_.Row(i),
+                           parent_->cplx_pre_.Row(i));
+  }
+
+  size_t block_length =
+      RealFourier::ComplexLength(RealFourier::FftOrder(num_frames));
+  RTC_CHECK_EQ(parent_->cplx_length_, block_length);
+  parent_->block_processor_->ProcessAudioBlock(
+      parent_->cplx_pre_.Array(), num_input_channels, parent_->cplx_length_,
+      num_output_channels, parent_->cplx_post_.Array());
+
+  for (size_t i = 0; i < num_output_channels; ++i) {
+    parent_->fft_->Inverse(parent_->cplx_post_.Row(i),
+                           parent_->real_buf_.Row(i));
+    memcpy(output[i], parent_->real_buf_.Row(i),
+           num_frames * sizeof(*input[0]));
+  }
+}
+
+LappedTransform::LappedTransform(size_t num_in_channels,
+                                 size_t num_out_channels,
+                                 size_t chunk_length,
+                                 const float* window,
+                                 size_t block_length,
+                                 size_t shift_amount,
+                                 Callback* callback)
+    : blocker_callback_(this),
+      num_in_channels_(num_in_channels),
+      num_out_channels_(num_out_channels),
+      block_length_(block_length),
+      chunk_length_(chunk_length),
+      block_processor_(callback),
+      blocker_(chunk_length_,
+               block_length_,
+               num_in_channels_,
+               num_out_channels_,
+               window,
+               shift_amount,
+               &blocker_callback_),
+      fft_(RealFourier::Create(RealFourier::FftOrder(block_length_))),
+      cplx_length_(RealFourier::ComplexLength(fft_->order())),
+      real_buf_(num_in_channels,
+                block_length_,
+                RealFourier::kFftBufferAlignment),
+      cplx_pre_(num_in_channels,
+                cplx_length_,
+                RealFourier::kFftBufferAlignment),
+      cplx_post_(num_out_channels,
+                 cplx_length_,
+                 RealFourier::kFftBufferAlignment) {
+  RTC_CHECK(num_in_channels_ > 0);
+  RTC_CHECK_GT(block_length_, 0);
+  RTC_CHECK_GT(chunk_length_, 0);
+  RTC_CHECK(block_processor_);
+
+  // block_length_ power of 2?
+  RTC_CHECK_EQ(0, block_length_ & (block_length_ - 1));
+}
+
+LappedTransform::~LappedTransform() = default;
+
+void LappedTransform::ProcessChunk(const float* const* in_chunk,
+                                   float* const* out_chunk) {
+  blocker_.ProcessChunk(in_chunk, chunk_length_, num_in_channels_,
+                        num_out_channels_, out_chunk);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h
new file mode 100644
index 0000000000..bb25c34a9e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h
@@ -0,0 +1,175 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
+
+#include <complex>
+#include <memory>
+
+#include "common_audio/real_fourier.h"
+#include "modules/audio_coding/codecs/opus/test/blocker.h"
+#include "rtc_base/memory/aligned_malloc.h"
+
+namespace webrtc {
+
+// Wrapper class for aligned arrays. Every row (and the first dimension) are
+// aligned to the given byte alignment.
+template <typename T>
+class AlignedArray {
+ public:
+  AlignedArray(size_t rows, size_t cols, size_t alignment)
+      : rows_(rows), cols_(cols) {
+    RTC_CHECK_GT(alignment, 0);
+    head_row_ =
+        static_cast<T**>(AlignedMalloc(rows_ * sizeof(*head_row_), alignment));
+    for (size_t i = 0; i < rows_; ++i) {
+      head_row_[i] = static_cast<T*>(
+          AlignedMalloc(cols_ * sizeof(**head_row_), alignment));
+    }
+  }
+
+  ~AlignedArray() {
+    for (size_t i = 0; i < rows_; ++i) {
+      AlignedFree(head_row_[i]);
+    }
+    AlignedFree(head_row_);
+  }
+
+  T* const* Array() { return head_row_; }
+
+  const T* const* Array() const { return head_row_; }
+
+  T* Row(size_t row) {
+    RTC_CHECK_LE(row, rows_);
+    return head_row_[row];
+  }
+
+  const T* Row(size_t row) const {
+    RTC_CHECK_LE(row, rows_);
+    return head_row_[row];
+  }
+
+ private:
+  size_t rows_;
+  size_t cols_;
+  T** head_row_;
+};
+
+// Helper class for audio processing modules which operate on frequency domain
+// input derived from the windowed time domain audio stream.
+//
+// The input audio chunk is sliced into possibly overlapping blocks, multiplied
+// by a window and transformed with an FFT implementation. The transformed data
+// is supplied to the given callback for processing. The processed output is
+// then inverse transformed into the time domain and spliced back into a chunk
+// which constitutes the final output of this processing module.
+class LappedTransform {
+ public:
+  class Callback {
+   public:
+    virtual ~Callback() {}
+
+    virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
+                                   size_t num_in_channels,
+                                   size_t frames,
+                                   size_t num_out_channels,
+                                   std::complex<float>* const* out_block) = 0;
+  };
+
+  // Construct a transform instance. `chunk_length` is the number of samples in
+  // each channel. `window` defines the window, owned by the caller (a copy is
+  // made internally); `window` should have length equal to `block_length`.
+  // `block_length` defines the length of a block, in samples.
+  // `shift_amount` is in samples. `callback` is the caller-owned audio
+  // processing function called for each block of the input chunk.
+  LappedTransform(size_t num_in_channels,
+                  size_t num_out_channels,
+                  size_t chunk_length,
+                  const float* window,
+                  size_t block_length,
+                  size_t shift_amount,
+                  Callback* callback);
+  ~LappedTransform();
+
+  // Main audio processing helper method. Internally slices `in_chunk` into
+  // blocks, transforms them to frequency domain, calls the callback for each
+  // block and returns a de-blocked time domain chunk of audio through
+  // `out_chunk`. Both buffers are caller-owned.
+  void ProcessChunk(const float* const* in_chunk, float* const* out_chunk);
+
+  // Get the chunk length.
+  //
+  // The chunk length is the number of samples per channel that must be passed
+  // to ProcessChunk via the parameter in_chunk.
+  //
+  // Returns the same chunk_length passed to the LappedTransform constructor.
+  size_t chunk_length() const { return chunk_length_; }
+
+  // Get the number of input channels.
+  //
+  // This is the number of arrays that must be passed to ProcessChunk via
+  // in_chunk.
+  //
+  // Returns the same num_in_channels passed to the LappedTransform constructor.
+  size_t num_in_channels() const { return num_in_channels_; }
+
+  // Get the number of output channels.
+  //
+  // This is the number of arrays that must be passed to ProcessChunk via
+  // out_chunk.
+  //
+  // Returns the same num_out_channels passed to the LappedTransform
+  // constructor.
+  size_t num_out_channels() const { return num_out_channels_; }
+
+  // Returns the initial delay.
+  //
+  // This is the delay introduced by the `blocker_` to be able to get and return
+  // chunks of `chunk_length`, but process blocks of `block_length`.
+  size_t initial_delay() const { return blocker_.initial_delay(); }
+
+ private:
+  // Internal middleware callback, given to the blocker. Transforms each block
+  // and hands it over to the processing method given at construction time.
+  class BlockThunk : public BlockerCallback {
+   public:
+    explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
+
+    void ProcessBlock(const float* const* input,
+                      size_t num_frames,
+                      size_t num_input_channels,
+                      size_t num_output_channels,
+                      float* const* output) override;
+
+   private:
+    LappedTransform* const parent_;
+  } blocker_callback_;
+
+  const size_t num_in_channels_;
+  const size_t num_out_channels_;
+
+  const size_t block_length_;
+  const size_t chunk_length_;
+
+  Callback* const block_processor_;
+  Blocker blocker_;
+
+  // TODO(alessiob): Replace RealFourier with a different FFT library.
+  std::unique_ptr<RealFourier> fft_;
+  const size_t cplx_length_;
+  AlignedArray<float> real_buf_;
+  AlignedArray<std::complex<float> > cplx_pre_;
+  AlignedArray<std::complex<float> > cplx_post_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc
new file mode 100644
index 0000000000..1003ed52e5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstring>
+
+#include "test/gtest.h"
+
+using std::complex;
+
+namespace {
+
+class NoopCallback : public webrtc::LappedTransform::Callback {
+ public:
+  NoopCallback() : block_num_(0) {}
+
+  void ProcessAudioBlock(const complex<float>* const* in_block,
+                         size_t in_channels,
+                         size_t frames,
+                         size_t out_channels,
+                         complex<float>* const* out_block) override {
+    RTC_CHECK_EQ(in_channels, out_channels);
+    for (size_t i = 0; i < out_channels; ++i) {
+      memcpy(out_block[i], in_block[i], sizeof(**in_block) * frames);
+    }
+    ++block_num_;
+  }
+
+  size_t block_num() { return block_num_; }
+
+ private:
+  size_t block_num_;
+};
+
+class FftCheckerCallback : public webrtc::LappedTransform::Callback {
+ public:
+  FftCheckerCallback() : block_num_(0) {}
+
+  void ProcessAudioBlock(const complex<float>* const* in_block,
+                         size_t in_channels,
+                         size_t frames,
+                         size_t out_channels,
+                         complex<float>* const* out_block) override {
+    RTC_CHECK_EQ(in_channels, out_channels);
+
+    size_t full_length = (frames - 1) * 2;
+    ++block_num_;
+
+    if (block_num_ > 0) {
+      ASSERT_NEAR(in_block[0][0].real(), static_cast<float>(full_length),
+                  1e-5f);
+      ASSERT_NEAR(in_block[0][0].imag(), 0.0f, 1e-5f);
+      for (size_t i = 1; i < frames; ++i) {
+        ASSERT_NEAR(in_block[0][i].real(), 0.0f, 1e-5f);
+        ASSERT_NEAR(in_block[0][i].imag(), 0.0f, 1e-5f);
+      }
+    }
+  }
+
+  size_t block_num() { return block_num_; }
+
+ private:
+  size_t block_num_;
+};
+
+void SetFloatArray(float value, int rows, int cols, float* const* array) {
+  for (int i = 0; i < rows; ++i) {
+    for (int j = 0; j < cols; ++j) {
+      array[i][j] = value;
+    }
+  }
+}
+
+}  // namespace
+
+namespace webrtc {
+
+TEST(LappedTransformTest, Windowless) {
+  const size_t kChannels = 3;
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  const size_t kShiftAmount = 64;
+  NoopCallback noop;
+
+  // Rectangular window.
+  float window[kBlockLength];
+  std::fill(window, &window[kBlockLength], 1.0f);
+
+  LappedTransform trans(kChannels, kChannels, kChunkLength, window,
+                        kBlockLength, kShiftAmount, &noop);
+  float in_buffer[kChannels][kChunkLength];
+  float* in_chunk[kChannels];
+  float out_buffer[kChannels][kChunkLength];
+  float* out_chunk[kChannels];
+
+  in_chunk[0] = in_buffer[0];
+  in_chunk[1] = in_buffer[1];
+  in_chunk[2] = in_buffer[2];
+  out_chunk[0] = out_buffer[0];
+  out_chunk[1] = out_buffer[1];
+  out_chunk[2] = out_buffer[2];
+  SetFloatArray(2.0f, kChannels, kChunkLength, in_chunk);
+  SetFloatArray(-1.0f, kChannels, kChunkLength, out_chunk);
+
+  trans.ProcessChunk(in_chunk, out_chunk);
+
+  for (size_t i = 0; i < kChannels; ++i) {
+    for (size_t j = 0; j < kChunkLength; ++j) {
+      ASSERT_NEAR(out_chunk[i][j], 2.0f, 1e-5f);
+    }
+  }
+
+  ASSERT_EQ(kChunkLength / kBlockLength, noop.block_num());
+}
+
+TEST(LappedTransformTest, IdentityProcessor) {
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  const size_t kShiftAmount = 32;
+  NoopCallback noop;
+
+  // Identity window for |overlap = block_size / 2|.
+  float window[kBlockLength];
+  std::fill(window, &window[kBlockLength], std::sqrt(0.5f));
+
+  LappedTransform trans(1, 1, kChunkLength, window, kBlockLength, kShiftAmount,
+                        &noop);
+  float in_buffer[kChunkLength];
+  float* in_chunk = in_buffer;
+  float out_buffer[kChunkLength];
+  float* out_chunk = out_buffer;
+
+  SetFloatArray(2.0f, 1, kChunkLength, &in_chunk);
+  SetFloatArray(-1.0f, 1, kChunkLength, &out_chunk);
+
+  trans.ProcessChunk(&in_chunk, &out_chunk);
+
+  for (size_t i = 0; i < kChunkLength; ++i) {
+    ASSERT_NEAR(out_chunk[i], (i < kBlockLength - kShiftAmount) ? 0.0f : 2.0f,
+                1e-5f);
+  }
+
+  ASSERT_EQ(kChunkLength / kShiftAmount, noop.block_num());
+}
+
+TEST(LappedTransformTest, Callbacks) {
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  FftCheckerCallback call;
+
+  // Rectangular window.
+  float window[kBlockLength];
+  std::fill(window, &window[kBlockLength], 1.0f);
+
+  LappedTransform trans(1, 1, kChunkLength, window, kBlockLength, kBlockLength,
+                        &call);
+  float in_buffer[kChunkLength];
+  float* in_chunk = in_buffer;
+  float out_buffer[kChunkLength];
+  float* out_chunk = out_buffer;
+
+  SetFloatArray(1.0f, 1, kChunkLength, &in_chunk);
+  SetFloatArray(-1.0f, 1, kChunkLength, &out_chunk);
+
+  trans.ProcessChunk(&in_chunk, &out_chunk);
+
+  ASSERT_EQ(kChunkLength / kBlockLength, call.block_num());
+}
+
+TEST(LappedTransformTest, chunk_length) {
+  const size_t kBlockLength = 64;
+  FftCheckerCallback call;
+  const float window[kBlockLength] = {};
+
+  // Make sure that chunk_length returns the same value passed to the
+  // LappedTransform constructor.
+  {
+    const size_t kExpectedChunkLength = 512;
+    const LappedTransform trans(1, 1, kExpectedChunkLength, window,
+                                kBlockLength, kBlockLength, &call);
+
+    EXPECT_EQ(kExpectedChunkLength, trans.chunk_length());
+  }
+  {
+    const size_t kExpectedChunkLength = 160;
+    const LappedTransform trans(1, 1, kExpectedChunkLength, window,
+                                kBlockLength, kBlockLength, &call);
+
+    EXPECT_EQ(kExpectedChunkLength, trans.chunk_length());
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
new file mode 100644
index 0000000000..7761efe8b3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+
+#include <utility>
+
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderPcm16B::AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels)
+    : sample_rate_hz_(sample_rate_hz), num_channels_(num_channels) {
+  RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+             sample_rate_hz == 32000 || sample_rate_hz == 48000)
+      << "Unsupported sample rate " << sample_rate_hz;
+  RTC_DCHECK_GE(num_channels, 1);
+}
+
+void AudioDecoderPcm16B::Reset() {}
+
+int AudioDecoderPcm16B::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
+size_t AudioDecoderPcm16B::Channels() const {
+  return num_channels_;
+}
+
+int AudioDecoderPcm16B::DecodeInternal(const uint8_t* encoded,
+                                       size_t encoded_len,
+                                       int sample_rate_hz,
+                                       int16_t* decoded,
+                                       SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz_, sample_rate_hz);
+  // Adjust the encoded length down to ensure the same number of samples in each
+  // channel.
+  const size_t encoded_len_adjusted =
+      PacketDuration(encoded, encoded_len) * 2 *
+      Channels();  // 2 bytes per sample per channel
+  size_t ret = WebRtcPcm16b_Decode(encoded, encoded_len_adjusted, decoded);
+  *speech_type = ConvertSpeechType(1);
+  return static_cast<int>(ret);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderPcm16B::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz_, 1000);
+  return LegacyEncodedAudioFrame::SplitBySamples(
+      this, std::move(payload), timestamp, samples_per_ms * 2 * num_channels_,
+      samples_per_ms);
+}
+
+int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
+                                       size_t encoded_len) const {
+  // Two encoded byte per sample per channel.
+  return static_cast<int>(encoded_len / (2 * Channels()));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
new file mode 100644
index 0000000000..6f50161d3f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoderPcm16B final : public AudioDecoder {
+ public:
+  AudioDecoderPcm16B(int sample_rate_hz, size_t num_channels);
+
+  AudioDecoderPcm16B(const AudioDecoderPcm16B&) = delete;
+  AudioDecoderPcm16B& operator=(const AudioDecoderPcm16B&) = delete;
+
+  void Reset() override;
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+ private:
+  const int sample_rate_hz_;
+  const size_t num_channels_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
new file mode 100644
index 0000000000..9445b1ee3e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
+                                      size_t input_len,
+                                      uint8_t* encoded) {
+  return WebRtcPcm16b_Encode(audio, input_len, encoded);
+}
+
+size_t AudioEncoderPcm16B::BytesPerSample() const {
+  return 2;
+}
+
+AudioEncoder::CodecType AudioEncoderPcm16B::GetCodecType() const {
+  return CodecType::kOther;
+}
+
+bool AudioEncoderPcm16B::Config::IsOk() const {
+  if ((sample_rate_hz != 8000) && (sample_rate_hz != 16000) &&
+      (sample_rate_hz != 32000) && (sample_rate_hz != 48000))
+    return false;
+  return AudioEncoderPcm::Config::IsOk();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
new file mode 100644
index 0000000000..c363b40b3f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
+
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+
+namespace webrtc {
+
+class AudioEncoderPcm16B final : public AudioEncoderPcm {
+ public:
+  struct Config : public AudioEncoderPcm::Config {
+   public:
+    Config() : AudioEncoderPcm::Config(107), sample_rate_hz(8000) {}
+    bool IsOk() const;
+
+    int sample_rate_hz;
+  };
+
+  explicit AudioEncoderPcm16B(const Config& config)
+      : AudioEncoderPcm(config, config.sample_rate_hz) {}
+
+  AudioEncoderPcm16B(const AudioEncoderPcm16B&) = delete;
+  AudioEncoderPcm16B& operator=(const AudioEncoderPcm16B&) = delete;
+
+ protected:
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
+
+  size_t BytesPerSample() const override;
+
+  AudioEncoder::CodecType GetCodecType() const override;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
new file mode 100644
index 0000000000..2f6dce5f41
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+
+size_t WebRtcPcm16b_Encode(const int16_t* speech,
+                           size_t len,
+                           uint8_t* encoded) {
+  size_t i;
+  for (i = 0; i < len; ++i) {
+    uint16_t s = speech[i];
+    encoded[2 * i] = s >> 8;
+    encoded[2 * i + 1] = s;
+  }
+  return 2 * len;
+}
+
+size_t WebRtcPcm16b_Decode(const uint8_t* encoded,
+                           size_t len,
+                           int16_t* speech) {
+  size_t i;
+  for (i = 0; i < len / 2; ++i)
+    speech[i] = encoded[2 * i] << 8 | encoded[2 * i + 1];
+  return len / 2;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
new file mode 100644
index 0000000000..75d1efda3b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
+/*
+ * Define the fixpoint numeric formats
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/****************************************************************************
+ * WebRtcPcm16b_Encode(...)
+ *
+ * "Encode" a sample vector to 16 bit linear (Encoded standard is big endian)
+ *
+ * Input:
+ *              - speech        : Input speech vector
+ *              - len           : Number of samples in speech vector
+ *
+ * Output:
+ *              - encoded       : Encoded data vector (big endian 16 bit)
+ *
+ * Returned value               : Length (in bytes) of coded data.
+ *                                Always equal to twice the len input parameter.
+ */
+
+size_t WebRtcPcm16b_Encode(const int16_t* speech, size_t len, uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcPcm16b_Decode(...)
+ *
+ * "Decode" a vector to 16 bit linear (Encoded standard is big endian)
+ *
+ * Input:
+ *              - encoded       : Encoded data vector (big endian 16 bit)
+ *              - len           : Number of bytes in encoded
+ *
+ * Output:
+ *              - speech        : Decoded speech vector
+ *
+ * Returned value               : Samples in speech
+ */
+
+size_t WebRtcPcm16b_Decode(const uint8_t* encoded, size_t len, int16_t* speech);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_ */
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
new file mode 100644
index 0000000000..ecf91b45ac
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
+
+#include <stdint.h>
+
+#include <initializer_list>
+
+namespace webrtc {
+
+void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs) {
+  for (uint8_t num_channels : {1, 2}) {
+    for (int sample_rate_hz : {8000, 16000, 32000}) {
+      specs->push_back(
+          {{"L16", sample_rate_hz, num_channels},
+           {sample_rate_hz, num_channels, sample_rate_hz * num_channels * 16}});
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h
new file mode 100644
index 0000000000..3fae717ff3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
+
+#include <vector>
+
+#include "api/audio_codecs/audio_format.h"
+
+namespace webrtc {
+void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs);
+}
+
+#endif  // MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
new file mode 100644
index 0000000000..724bba52d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -0,0 +1,272 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+
+#include <string.h>
+
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/byte_order.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+static constexpr const int kRedMaxPacketSize =
+    1 << 10;  // RED packets must be less than 1024 bytes to fit the 10 bit
+              // block length.
+static constexpr const size_t kRedMaxTimestampDelta =
+    1 << 14;  // RED packets can encode a timestamp delta of 14 bits.
+static constexpr const size_t kAudioMaxRtpPacketLen =
+    1200;  // The typical MTU is 1200 bytes.
+
+static constexpr size_t kRedHeaderLength = 4;  // 4 bytes RED header.
+static constexpr size_t kRedLastHeaderLength =
+    1;  // reduced size for last RED header.
+
+static constexpr size_t kRedNumberOfRedundantEncodings =
+    1;  // The level of redundancy we support.
+
+AudioEncoderCopyRed::Config::Config() = default;
+AudioEncoderCopyRed::Config::Config(Config&&) = default;
+AudioEncoderCopyRed::Config::~Config() = default;
+
+size_t GetMaxRedundancyFromFieldTrial(const FieldTrialsView& field_trials) {
+  const std::string red_trial =
+      field_trials.Lookup("WebRTC-Audio-Red-For-Opus");
+  size_t redundancy = 0;
+  if (sscanf(red_trial.c_str(), "Enabled-%zu", &redundancy) != 1 ||
+      redundancy > 9) {
+    return kRedNumberOfRedundantEncodings;
+  }
+  return redundancy;
+}
+
+AudioEncoderCopyRed::AudioEncoderCopyRed(Config&& config,
+                                         const FieldTrialsView& field_trials)
+    : speech_encoder_(std::move(config.speech_encoder)),
+      primary_encoded_(0, kAudioMaxRtpPacketLen),
+      max_packet_length_(kAudioMaxRtpPacketLen),
+      red_payload_type_(config.payload_type) {
+  RTC_CHECK(speech_encoder_) << "Speech encoder not provided.";
+
+  auto number_of_redundant_encodings =
+      GetMaxRedundancyFromFieldTrial(field_trials);
+  for (size_t i = 0; i < number_of_redundant_encodings; i++) {
+    std::pair<EncodedInfo, rtc::Buffer> redundant;
+    redundant.second.EnsureCapacity(kAudioMaxRtpPacketLen);
+    redundant_encodings_.push_front(std::move(redundant));
+  }
+}
+
+AudioEncoderCopyRed::~AudioEncoderCopyRed() = default;
+
+int AudioEncoderCopyRed::SampleRateHz() const {
+  return speech_encoder_->SampleRateHz();
+}
+
+size_t AudioEncoderCopyRed::NumChannels() const {
+  return speech_encoder_->NumChannels();
+}
+
+int AudioEncoderCopyRed::RtpTimestampRateHz() const {
+  return speech_encoder_->RtpTimestampRateHz();
+}
+
+size_t AudioEncoderCopyRed::Num10MsFramesInNextPacket() const {
+  return speech_encoder_->Num10MsFramesInNextPacket();
+}
+
+size_t AudioEncoderCopyRed::Max10MsFramesInAPacket() const {
+  return speech_encoder_->Max10MsFramesInAPacket();
+}
+
+int AudioEncoderCopyRed::GetTargetBitrate() const {
+  return speech_encoder_->GetTargetBitrate();
+}
+
+AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  primary_encoded_.Clear();
+  EncodedInfo info =
+      speech_encoder_->Encode(rtp_timestamp, audio, &primary_encoded_);
+  RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
+  RTC_DCHECK_EQ(primary_encoded_.size(), info.encoded_bytes);
+
+  if (info.encoded_bytes == 0 || info.encoded_bytes >= kRedMaxPacketSize) {
+    return info;
+  }
+  RTC_DCHECK_GT(max_packet_length_, info.encoded_bytes);
+
+  size_t header_length_bytes = kRedLastHeaderLength;
+  size_t bytes_available = max_packet_length_ - info.encoded_bytes;
+  auto it = redundant_encodings_.begin();
+
+  // Determine how much redundancy we can fit into our packet by
+  // iterating forward. This is determined both by the length as well
+  // as the timestamp difference. The latter can occur with opus DTX which
+  // has timestamp gaps of 400ms which exceeds REDs timestamp delta field size.
+  for (; it != redundant_encodings_.end(); it++) {
+    if (bytes_available < kRedHeaderLength + it->first.encoded_bytes) {
+      break;
+    }
+    if (it->first.encoded_bytes == 0) {
+      break;
+    }
+    if (rtp_timestamp - it->first.encoded_timestamp >= kRedMaxTimestampDelta) {
+      break;
+    }
+    bytes_available -= kRedHeaderLength + it->first.encoded_bytes;
+    header_length_bytes += kRedHeaderLength;
+  }
+
+  // Allocate room for RFC 2198 header.
+  encoded->SetSize(header_length_bytes);
+
+  // Iterate backwards and append the data.
+  size_t header_offset = 0;
+  while (it-- != redundant_encodings_.begin()) {
+    encoded->AppendData(it->second);
+
+    const uint32_t timestamp_delta =
+        info.encoded_timestamp - it->first.encoded_timestamp;
+    encoded->data()[header_offset] = it->first.payload_type | 0x80;
+    rtc::SetBE16(static_cast<uint8_t*>(encoded->data()) + header_offset + 1,
+                 (timestamp_delta << 2) | (it->first.encoded_bytes >> 8));
+    encoded->data()[header_offset + 3] = it->first.encoded_bytes & 0xff;
+    header_offset += kRedHeaderLength;
+    info.redundant.push_back(it->first);
+  }
+
+  // `info` will be implicitly cast to an EncodedInfoLeaf struct, effectively
+  // discarding the (empty) vector of redundant information. This is
+  // intentional.
+  if (header_length_bytes > kRedHeaderLength) {
+    info.redundant.push_back(info);
+    RTC_DCHECK_EQ(info.speech,
+                  info.redundant[info.redundant.size() - 1].speech);
+  }
+
+  encoded->AppendData(primary_encoded_);
+  RTC_DCHECK_EQ(header_offset, header_length_bytes - 1);
+  encoded->data()[header_offset] = info.payload_type;
+
+  // Shift the redundant encodings.
+  auto rit = redundant_encodings_.rbegin();
+  for (auto next = std::next(rit); next != redundant_encodings_.rend();
+       rit++, next = std::next(rit)) {
+    rit->first = next->first;
+    rit->second.SetData(next->second);
+  }
+  it = redundant_encodings_.begin();
+  if (it != redundant_encodings_.end()) {
+    it->first = info;
+    it->second.SetData(primary_encoded_);
+  }
+
+  // Update main EncodedInfo.
+  info.payload_type = red_payload_type_;
+  info.encoded_bytes = encoded->size();
+  return info;
+}
+
+void AudioEncoderCopyRed::Reset() {
+  speech_encoder_->Reset();
+  auto number_of_redundant_encodings = redundant_encodings_.size();
+  redundant_encodings_.clear();
+  for (size_t i = 0; i < number_of_redundant_encodings; i++) {
+    std::pair<EncodedInfo, rtc::Buffer> redundant;
+    redundant.second.EnsureCapacity(kAudioMaxRtpPacketLen);
+    redundant_encodings_.push_front(std::move(redundant));
+  }
+}
+
+bool AudioEncoderCopyRed::SetFec(bool enable) {
+  return speech_encoder_->SetFec(enable);
+}
+
+bool AudioEncoderCopyRed::SetDtx(bool enable) {
+  return speech_encoder_->SetDtx(enable);
+}
+
+bool AudioEncoderCopyRed::GetDtx() const {
+  return speech_encoder_->GetDtx();
+}
+
+bool AudioEncoderCopyRed::SetApplication(Application application) {
+  return speech_encoder_->SetApplication(application);
+}
+
+void AudioEncoderCopyRed::SetMaxPlaybackRate(int frequency_hz) {
+  speech_encoder_->SetMaxPlaybackRate(frequency_hz);
+}
+
+bool AudioEncoderCopyRed::EnableAudioNetworkAdaptor(
+    const std::string& config_string,
+    RtcEventLog* event_log) {
+  return speech_encoder_->EnableAudioNetworkAdaptor(config_string, event_log);
+}
+
+void AudioEncoderCopyRed::DisableAudioNetworkAdaptor() {
+  speech_encoder_->DisableAudioNetworkAdaptor();
+}
+
+void AudioEncoderCopyRed::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  speech_encoder_->OnReceivedUplinkPacketLossFraction(
+      uplink_packet_loss_fraction);
+}
+
+void AudioEncoderCopyRed::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {
+  speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
+                                             bwe_period_ms);
+}
+
+void AudioEncoderCopyRed::OnReceivedUplinkAllocation(
+    BitrateAllocationUpdate update) {
+  speech_encoder_->OnReceivedUplinkAllocation(update);
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderCopyRed::GetFrameLengthRange() const {
+  return speech_encoder_->GetFrameLengthRange();
+}
+
+void AudioEncoderCopyRed::OnReceivedRtt(int rtt_ms) {
+  speech_encoder_->OnReceivedRtt(rtt_ms);
+}
+
+void AudioEncoderCopyRed::OnReceivedOverhead(size_t overhead_bytes_per_packet) {
+  max_packet_length_ = kAudioMaxRtpPacketLen - overhead_bytes_per_packet;
+  return speech_encoder_->OnReceivedOverhead(overhead_bytes_per_packet);
+}
+
+void AudioEncoderCopyRed::SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                                      int max_frame_length_ms) {
+  return speech_encoder_->SetReceiverFrameLengthRange(min_frame_length_ms,
+                                                      max_frame_length_ms);
+}
+
+ANAStats AudioEncoderCopyRed::GetANAStats() const {
+  return speech_encoder_->GetANAStats();
+}
+
+rtc::ArrayView<std::unique_ptr<AudioEncoder>>
+AudioEncoderCopyRed::ReclaimContainedEncoders() {
+  return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
new file mode 100644
index 0000000000..359b5eaa17
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
+#define MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/field_trials_view.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+// This class implements redundant audio coding as described in
+//   https://tools.ietf.org/html/rfc2198
+// The class object will have an underlying AudioEncoder object that performs
+// the actual encodings. The current class will gather the N latest encodings
+// from the underlying codec into one packet. Currently N is hard-coded to 2.
+
+class AudioEncoderCopyRed final : public AudioEncoder {
+ public:
+  struct Config {
+    Config();
+    Config(Config&&);
+    ~Config();
+    int payload_type;
+    std::unique_ptr<AudioEncoder> speech_encoder;
+  };
+
+  AudioEncoderCopyRed(Config&& config, const FieldTrialsView& field_trials);
+
+  ~AudioEncoderCopyRed() override;
+
+  AudioEncoderCopyRed(const AudioEncoderCopyRed&) = delete;
+  AudioEncoderCopyRed& operator=(const AudioEncoderCopyRed&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  void Reset() override;
+  bool SetFec(bool enable) override;
+
+  bool SetDtx(bool enable) override;
+  bool GetDtx() const override;
+
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  bool EnableAudioNetworkAdaptor(const std::string& config_string,
+                                 RtcEventLog* event_log) override;
+  void DisableAudioNetworkAdaptor() override;
+  void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) override;
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms) override;
+  void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
+  void OnReceivedRtt(int rtt_ms) override;
+  void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
+  void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                   int max_frame_length_ms) override;
+  ANAStats GetANAStats() const override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+  rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
+      override;
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  std::unique_ptr<AudioEncoder> speech_encoder_;
+  rtc::Buffer primary_encoded_;
+  size_t max_packet_length_;
+  int red_payload_type_;
+  std::list<std::pair<EncodedInfo, rtc::Buffer>> redundant_encodings_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
new file mode 100644
index 0000000000..795a996624
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -0,0 +1,641 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+
+#include <memory>
+#include <vector>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
+#include "test/scoped_key_value_config.h"
+#include "test/testsupport/rtc_expect_death.h"
+
+using ::testing::_;
+using ::testing::Eq;
+using ::testing::InSequence;
+using ::testing::Invoke;
+using ::testing::MockFunction;
+using ::testing::Not;
+using ::testing::Optional;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+namespace webrtc {
+
+namespace {
+static const size_t kMaxNumSamples = 48 * 10 * 2;  // 10 ms @ 48 kHz stereo.
+static const size_t kRedLastHeaderLength =
+    1;  // 1 byte RED header for the last element.
+}
+
+class AudioEncoderCopyRedTest : public ::testing::Test {
+ protected:
+  AudioEncoderCopyRedTest()
+      : mock_encoder_(new MockAudioEncoder),
+        timestamp_(4711),
+        sample_rate_hz_(16000),
+        num_audio_samples_10ms(sample_rate_hz_ / 100),
+        red_payload_type_(200) {
+    AudioEncoderCopyRed::Config config;
+    config.payload_type = red_payload_type_;
+    config.speech_encoder = std::unique_ptr<AudioEncoder>(mock_encoder_);
+    red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials_));
+    memset(audio_, 0, sizeof(audio_));
+    EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(1U));
+    EXPECT_CALL(*mock_encoder_, SampleRateHz())
+        .WillRepeatedly(Return(sample_rate_hz_));
+  }
+
+  void TearDown() override { red_.reset(); }
+
+  void Encode() {
+    ASSERT_TRUE(red_.get() != NULL);
+    encoded_.Clear();
+    encoded_info_ = red_->Encode(
+        timestamp_,
+        rtc::ArrayView<const int16_t>(audio_, num_audio_samples_10ms),
+        &encoded_);
+    timestamp_ += rtc::checked_cast<uint32_t>(num_audio_samples_10ms);
+  }
+
+  test::ScopedKeyValueConfig field_trials_;
+  MockAudioEncoder* mock_encoder_;
+  std::unique_ptr<AudioEncoderCopyRed> red_;
+  uint32_t timestamp_;
+  int16_t audio_[kMaxNumSamples];
+  const int sample_rate_hz_;
+  size_t num_audio_samples_10ms;
+  rtc::Buffer encoded_;
+  AudioEncoder::EncodedInfo encoded_info_;
+  const int red_payload_type_;
+};
+
+TEST_F(AudioEncoderCopyRedTest, CreateAndDestroy) {}
+
+TEST_F(AudioEncoderCopyRedTest, CheckSampleRatePropagation) {
+  EXPECT_CALL(*mock_encoder_, SampleRateHz()).WillOnce(Return(17));
+  EXPECT_EQ(17, red_->SampleRateHz());
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckNumChannelsPropagation) {
+  EXPECT_CALL(*mock_encoder_, NumChannels()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->NumChannels());
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckFrameSizePropagation) {
+  EXPECT_CALL(*mock_encoder_, Num10MsFramesInNextPacket())
+      .WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->Num10MsFramesInNextPacket());
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckMaxFrameSizePropagation) {
+  EXPECT_CALL(*mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->Max10MsFramesInAPacket());
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) {
+  EXPECT_CALL(*mock_encoder_,
+              OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
+  red_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) {
+  EXPECT_CALL(*mock_encoder_, OnReceivedUplinkPacketLossFraction(0.5));
+  red_->OnReceivedUplinkPacketLossFraction(0.5);
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckGetFrameLengthRangePropagation) {
+  auto expected_range =
+      std::make_pair(TimeDelta::Millis(20), TimeDelta::Millis(20));
+  EXPECT_CALL(*mock_encoder_, GetFrameLengthRange())
+      .WillRepeatedly(Return(absl::make_optional(expected_range)));
+  EXPECT_THAT(red_->GetFrameLengthRange(), Optional(Eq(expected_range)));
+}
+
+// Checks that the an Encode() call is immediately propagated to the speech
+// encoder.
+TEST_F(AudioEncoderCopyRedTest, CheckImmediateEncode) {
+  // Interleaving the EXPECT_CALL sequence with expectations on the MockFunction
+  // check ensures that exactly one call to EncodeImpl happens in each
+  // Encode call.
+  InSequence s;
+  MockFunction<void(int check_point_id)> check;
+  for (int i = 1; i <= 6; ++i) {
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillRepeatedly(Return(AudioEncoder::EncodedInfo()));
+    EXPECT_CALL(check, Call(i));
+    Encode();
+    check.Call(i);
+  }
+}
+
+// Checks that no output is produced if the underlying codec doesn't emit any
+// new data, even if the RED codec is loaded with a secondary encoding.
+TEST_F(AudioEncoderCopyRedTest, CheckNoOutput) {
+  static const size_t kEncodedSize = 17;
+  static const size_t kHeaderLenBytes = 5;
+  {
+    InSequence s;
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(kEncodedSize)))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(0)))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(kEncodedSize)));
+  }
+
+  // Start with one Encode() call that will produce output.
+  Encode();
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  EXPECT_EQ(0u, encoded_info_.redundant.size());
+  EXPECT_EQ(kEncodedSize + kRedLastHeaderLength, encoded_info_.encoded_bytes);
+
+  // Next call to the speech encoder will not produce any output.
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.encoded_bytes);
+
+  // Final call to the speech encoder will produce output.
+  Encode();
+  EXPECT_EQ(2 * kEncodedSize + kHeaderLenBytes, encoded_info_.encoded_bytes);
+  ASSERT_EQ(2u, encoded_info_.redundant.size());
+}
+
+// Checks that the correct payload sizes are populated into the redundancy
+// information for a redundancy level of 1.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes1) {
+  // Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
+  // of calls.
+  static const int kNumPackets = 10;
+  InSequence s;
+  for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) {
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(encode_size)));
+  }
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.redundant.size());
+  EXPECT_EQ(kRedLastHeaderLength + 1u, encoded_info_.encoded_bytes);
+
+  for (size_t i = 2; i <= kNumPackets; ++i) {
+    Encode();
+    ASSERT_EQ(2u, encoded_info_.redundant.size());
+    EXPECT_EQ(i, encoded_info_.redundant[1].encoded_bytes);
+    EXPECT_EQ(i - 1, encoded_info_.redundant[0].encoded_bytes);
+    EXPECT_EQ(5 + i + (i - 1), encoded_info_.encoded_bytes);
+  }
+}
+
+// Checks that the correct payload sizes are populated into the redundancy
+// information for a redundancy level of 0.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes0) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-0/");
+  // Recreate the RED encoder to take the new field trial setting into account.
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type_;
+  config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
+  red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
+
+  // Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
+  // of calls.
+  static const int kNumPackets = 10;
+  InSequence s;
+  for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) {
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(encode_size)));
+  }
+
+  for (size_t i = 1; i <= kNumPackets; ++i) {
+    Encode();
+    ASSERT_EQ(0u, encoded_info_.redundant.size());
+    EXPECT_EQ(1 + i, encoded_info_.encoded_bytes);
+  }
+}
+// Checks that the correct payload sizes are populated into the redundancy
+// information for a redundancy level of 2.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes2) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-2/");
+  // Recreate the RED encoder to take the new field trial setting into account.
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type_;
+  config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
+  red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
+
+  // Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
+  // of calls.
+  static const int kNumPackets = 10;
+  InSequence s;
+  for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) {
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(encode_size)));
+  }
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.redundant.size());
+  EXPECT_EQ(kRedLastHeaderLength + 1u, encoded_info_.encoded_bytes);
+
+  // Second call is also special since it does not include a tertiary
+  // payload.
+  Encode();
+  EXPECT_EQ(2u, encoded_info_.redundant.size());
+  EXPECT_EQ(8u, encoded_info_.encoded_bytes);
+
+  for (size_t i = 3; i <= kNumPackets; ++i) {
+    Encode();
+    ASSERT_EQ(3u, encoded_info_.redundant.size());
+    EXPECT_EQ(i, encoded_info_.redundant[2].encoded_bytes);
+    EXPECT_EQ(i - 1, encoded_info_.redundant[1].encoded_bytes);
+    EXPECT_EQ(i - 2, encoded_info_.redundant[0].encoded_bytes);
+    EXPECT_EQ(9 + i + (i - 1) + (i - 2), encoded_info_.encoded_bytes);
+  }
+}
+
+// Checks that the correct payload sizes are populated into the redundancy
+// information for a redundancy level of 3.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloadSizes3) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-3/");
+  // Recreate the RED encoder to take the new field trial setting into account.
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type_;
+  config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
+  red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials_));
+
+  // Let the mock encoder return payload sizes 1, 2, 3, ..., 10 for the sequence
+  // of calls.
+  static const int kNumPackets = 10;
+  InSequence s;
+  for (int encode_size = 1; encode_size <= kNumPackets; ++encode_size) {
+    EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+        .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(encode_size)));
+  }
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  EXPECT_EQ(0u, encoded_info_.redundant.size());
+  EXPECT_EQ(kRedLastHeaderLength + 1u, encoded_info_.encoded_bytes);
+
+  // Second call is also special since it does not include a tertiary
+  // payload.
+  Encode();
+  EXPECT_EQ(2u, encoded_info_.redundant.size());
+  EXPECT_EQ(8u, encoded_info_.encoded_bytes);
+
+  // Third call is also special since it does not include a quaternary
+  // payload.
+  Encode();
+  EXPECT_EQ(3u, encoded_info_.redundant.size());
+  EXPECT_EQ(15u, encoded_info_.encoded_bytes);
+
+  for (size_t i = 4; i <= kNumPackets; ++i) {
+    Encode();
+    ASSERT_EQ(4u, encoded_info_.redundant.size());
+    EXPECT_EQ(i, encoded_info_.redundant[3].encoded_bytes);
+    EXPECT_EQ(i - 1, encoded_info_.redundant[2].encoded_bytes);
+    EXPECT_EQ(i - 2, encoded_info_.redundant[1].encoded_bytes);
+    EXPECT_EQ(i - 3, encoded_info_.redundant[0].encoded_bytes);
+    EXPECT_EQ(13 + i + (i - 1) + (i - 2) + (i - 3),
+              encoded_info_.encoded_bytes);
+  }
+}
+
+// Checks that the correct timestamps are returned.
+TEST_F(AudioEncoderCopyRedTest, CheckTimestamps) {
+  uint32_t primary_timestamp = timestamp_;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 17;
+  info.encoded_timestamp = timestamp_;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  EXPECT_EQ(primary_timestamp, encoded_info_.encoded_timestamp);
+
+  uint32_t secondary_timestamp = primary_timestamp;
+  primary_timestamp = timestamp_;
+  info.encoded_timestamp = timestamp_;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+
+  Encode();
+  ASSERT_EQ(2u, encoded_info_.redundant.size());
+  EXPECT_EQ(primary_timestamp, encoded_info_.redundant[1].encoded_timestamp);
+  EXPECT_EQ(secondary_timestamp, encoded_info_.redundant[0].encoded_timestamp);
+  EXPECT_EQ(primary_timestamp, encoded_info_.encoded_timestamp);
+}
+
+// Checks that the primary and secondary payloads are written correctly.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloads) {
+  // Let the mock encoder write payloads with increasing values. The first
+  // payload will have values 0, 1, 2, ..., kPayloadLenBytes - 1.
+  static const size_t kPayloadLenBytes = 5;
+  static const size_t kHeaderLenBytes = 5;
+  uint8_t payload[kPayloadLenBytes];
+  for (uint8_t i = 0; i < kPayloadLenBytes; ++i) {
+    payload[i] = i;
+  }
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillRepeatedly(Invoke(MockAudioEncoder::CopyEncoding(payload)));
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  EXPECT_EQ(kRedLastHeaderLength + kPayloadLenBytes,
+            encoded_info_.encoded_bytes);
+  for (size_t i = 0; i < kPayloadLenBytes; ++i) {
+    EXPECT_EQ(i, encoded_.data()[kRedLastHeaderLength + i]);
+  }
+
+  for (int j = 0; j < 1; ++j) {
+    // Increment all values of the payload by 10.
+    for (size_t i = 0; i < kPayloadLenBytes; ++i)
+      payload[i] += 10;
+
+    Encode();
+    ASSERT_EQ(2u, encoded_info_.redundant.size());
+    EXPECT_EQ(kPayloadLenBytes, encoded_info_.redundant[0].encoded_bytes);
+    EXPECT_EQ(kPayloadLenBytes, encoded_info_.redundant[1].encoded_bytes);
+    for (size_t i = 0; i < kPayloadLenBytes; ++i) {
+      // Check secondary payload.
+      EXPECT_EQ(j * 10 + i, encoded_.data()[kHeaderLenBytes + i]);
+
+      // Check primary payload.
+      EXPECT_EQ((j + 1) * 10 + i,
+                encoded_.data()[kHeaderLenBytes + i + kPayloadLenBytes]);
+    }
+  }
+}
+
+// Checks correct propagation of payload type.
+TEST_F(AudioEncoderCopyRedTest, CheckPayloadType) {
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 17;
+  info.payload_type = primary_payload_type;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+
+  // First call is a special case, since it does not include a secondary
+  // payload.
+  Encode();
+  ASSERT_EQ(0u, encoded_info_.redundant.size());
+
+  const int secondary_payload_type = red_payload_type_ + 2;
+  info.payload_type = secondary_payload_type;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+
+  Encode();
+  ASSERT_EQ(2u, encoded_info_.redundant.size());
+  EXPECT_EQ(secondary_payload_type, encoded_info_.redundant[1].payload_type);
+  EXPECT_EQ(primary_payload_type, encoded_info_.redundant[0].payload_type);
+  EXPECT_EQ(red_payload_type_, encoded_info_.payload_type);
+}
+
+TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header) {
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 10;
+  info.encoded_timestamp = timestamp_;
+  info.payload_type = primary_payload_type;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Second call will produce a redundant encoding.
+
+  EXPECT_EQ(encoded_.size(),
+            5u + 2 * 10u);  // header size + two encoded payloads.
+  EXPECT_EQ(encoded_[0], primary_payload_type | 0x80);
+
+  uint32_t timestamp_delta = encoded_info_.encoded_timestamp -
+                             encoded_info_.redundant[0].encoded_timestamp;
+  // Timestamp delta is encoded as a 14 bit value.
+  EXPECT_EQ(encoded_[1], timestamp_delta >> 6);
+  EXPECT_EQ(static_cast<uint8_t>(encoded_[2] >> 2), timestamp_delta & 0x3f);
+  // Redundant length is encoded as 10 bit value.
+  EXPECT_EQ(encoded_[2] & 0x3u, encoded_info_.redundant[1].encoded_bytes >> 8);
+  EXPECT_EQ(encoded_[3], encoded_info_.redundant[1].encoded_bytes & 0xff);
+  EXPECT_EQ(encoded_[4], primary_payload_type);
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Third call will produce a redundant encoding with double
+             // redundancy.
+
+  EXPECT_EQ(encoded_.size(),
+            5u + 2 * 10u);  // header size + two encoded payloads.
+  EXPECT_EQ(encoded_[0], primary_payload_type | 0x80);
+
+  timestamp_delta = encoded_info_.encoded_timestamp -
+                    encoded_info_.redundant[0].encoded_timestamp;
+  // Timestamp delta is encoded as a 14 bit value.
+  EXPECT_EQ(encoded_[1], timestamp_delta >> 6);
+  EXPECT_EQ(static_cast<uint8_t>(encoded_[2] >> 2), timestamp_delta & 0x3f);
+  // Redundant length is encoded as 10 bit value.
+  EXPECT_EQ(encoded_[2] & 0x3u, encoded_info_.redundant[1].encoded_bytes >> 8);
+  EXPECT_EQ(encoded_[3], encoded_info_.redundant[1].encoded_bytes & 0xff);
+
+  EXPECT_EQ(encoded_[4], primary_payload_type | 0x80);
+  timestamp_delta = encoded_info_.encoded_timestamp -
+                    encoded_info_.redundant[1].encoded_timestamp;
+}
+
+// Variant with a redundancy of 0.
+TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header0) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-0/");
+  // Recreate the RED encoder to take the new field trial setting into account.
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type_;
+  config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
+  red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
+
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 10;
+  info.encoded_timestamp = timestamp_;
+  info.payload_type = primary_payload_type;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Second call will not produce a redundant encoding.
+
+  EXPECT_EQ(encoded_.size(),
+            1u + 1 * 10u);  // header size + one encoded payloads.
+  EXPECT_EQ(encoded_[0], primary_payload_type);
+}
+// Variant with a redundancy of 2.
+TEST_F(AudioEncoderCopyRedTest, CheckRFC2198Header2) {
+  webrtc::test::ScopedKeyValueConfig field_trials(
+      field_trials_, "WebRTC-Audio-Red-For-Opus/Enabled-2/");
+  // Recreate the RED encoder to take the new field trial setting into account.
+  AudioEncoderCopyRed::Config config;
+  config.payload_type = red_payload_type_;
+  config.speech_encoder = std::move(red_->ReclaimContainedEncoders()[0]);
+  red_.reset(new AudioEncoderCopyRed(std::move(config), field_trials));
+
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 10;
+  info.encoded_timestamp = timestamp_;
+  info.payload_type = primary_payload_type;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Second call will produce a redundant encoding.
+
+  EXPECT_EQ(encoded_.size(),
+            5u + 2 * 10u);  // header size + two encoded payloads.
+  EXPECT_EQ(encoded_[0], primary_payload_type | 0x80);
+
+  uint32_t timestamp_delta = encoded_info_.encoded_timestamp -
+                             encoded_info_.redundant[0].encoded_timestamp;
+  // Timestamp delta is encoded as a 14 bit value.
+  EXPECT_EQ(encoded_[1], timestamp_delta >> 6);
+  EXPECT_EQ(static_cast<uint8_t>(encoded_[2] >> 2), timestamp_delta & 0x3f);
+  // Redundant length is encoded as 10 bit value.
+  EXPECT_EQ(encoded_[2] & 0x3u, encoded_info_.redundant[1].encoded_bytes >> 8);
+  EXPECT_EQ(encoded_[3], encoded_info_.redundant[1].encoded_bytes & 0xff);
+  EXPECT_EQ(encoded_[4], primary_payload_type);
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Third call will produce a redundant encoding with double
+             // redundancy.
+
+  EXPECT_EQ(encoded_.size(),
+            9u + 3 * 10u);  // header size + three encoded payloads.
+  EXPECT_EQ(encoded_[0], primary_payload_type | 0x80);
+
+  timestamp_delta = encoded_info_.encoded_timestamp -
+                    encoded_info_.redundant[0].encoded_timestamp;
+  // Timestamp delta is encoded as a 14 bit value.
+  EXPECT_EQ(encoded_[1], timestamp_delta >> 6);
+  EXPECT_EQ(static_cast<uint8_t>(encoded_[2] >> 2), timestamp_delta & 0x3f);
+  // Redundant length is encoded as 10 bit value.
+  EXPECT_EQ(encoded_[2] & 0x3u, encoded_info_.redundant[1].encoded_bytes >> 8);
+  EXPECT_EQ(encoded_[3], encoded_info_.redundant[1].encoded_bytes & 0xff);
+
+  EXPECT_EQ(encoded_[4], primary_payload_type | 0x80);
+  timestamp_delta = encoded_info_.encoded_timestamp -
+                    encoded_info_.redundant[1].encoded_timestamp;
+  // Timestamp delta is encoded as a 14 bit value.
+  EXPECT_EQ(encoded_[5], timestamp_delta >> 6);
+  EXPECT_EQ(static_cast<uint8_t>(encoded_[6] >> 2), timestamp_delta & 0x3f);
+  // Redundant length is encoded as 10 bit value.
+  EXPECT_EQ(encoded_[6] & 0x3u, encoded_info_.redundant[1].encoded_bytes >> 8);
+  EXPECT_EQ(encoded_[7], encoded_info_.redundant[1].encoded_bytes & 0xff);
+  EXPECT_EQ(encoded_[8], primary_payload_type);
+}
+
+TEST_F(AudioEncoderCopyRedTest, RespectsPayloadMTU) {
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 600;
+  info.encoded_timestamp = timestamp_;
+  info.payload_type = primary_payload_type;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  info.encoded_bytes = 500;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Second call will produce a redundant encoding.
+
+  EXPECT_EQ(encoded_.size(), 5u + 600u + 500u);
+
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  info.encoded_bytes = 400;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();  // Third call will drop the oldest packet.
+  EXPECT_EQ(encoded_.size(), 5u + 500u + 400u);
+}
+
+TEST_F(AudioEncoderCopyRedTest, LargeTimestampGap) {
+  const int primary_payload_type = red_payload_type_ + 1;
+  AudioEncoder::EncodedInfo info;
+  info.encoded_bytes = 100;
+  info.encoded_timestamp = timestamp_;
+  info.payload_type = primary_payload_type;
+
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+  // Update timestamp to simulate a 400ms gap like the one
+  // opus DTX causes.
+  timestamp_ += 19200;
+  info.encoded_timestamp = timestamp_;  // update timestamp.
+  info.encoded_bytes = 200;
+  EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
+      .WillOnce(Invoke(MockAudioEncoder::FakeEncoding(info)));
+  Encode();
+
+  // The old packet will be dropped.
+  EXPECT_EQ(encoded_.size(), 1u + 200u);
+}
+
+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// This test fixture tests various error conditions that makes the
+// AudioEncoderCng die via CHECKs.
+class AudioEncoderCopyRedDeathTest : public AudioEncoderCopyRedTest {
+ protected:
+  AudioEncoderCopyRedDeathTest() : AudioEncoderCopyRedTest() {}
+};
+
+TEST_F(AudioEncoderCopyRedDeathTest, WrongFrameSize) {
+  num_audio_samples_10ms *= 2;  // 20 ms frame.
+  RTC_EXPECT_DEATH(Encode(), "");
+  num_audio_samples_10ms = 0;  // Zero samples.
+  RTC_EXPECT_DEATH(Encode(), "");
+}
+
+TEST_F(AudioEncoderCopyRedDeathTest, NullSpeechEncoder) {
+  test::ScopedKeyValueConfig field_trials;
+  AudioEncoderCopyRed* red = NULL;
+  AudioEncoderCopyRed::Config config;
+  config.speech_encoder = NULL;
+  RTC_EXPECT_DEATH(
+      red = new AudioEncoderCopyRed(std::move(config), field_trials),
+      "Speech encoder not provided.");
+  // The delete operation is needed to avoid leak reports from memcheck.
+  delete red;
+}
+
+#endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
new file mode 100644
index 0000000000..537e6fcede
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
+
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using ::std::get;
+
+namespace webrtc {
+
+AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
+                                         int input_sampling_khz,
+                                         int output_sampling_khz)
+    : block_duration_ms_(block_duration_ms),
+      input_sampling_khz_(input_sampling_khz),
+      output_sampling_khz_(output_sampling_khz),
+      input_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * input_sampling_khz_)),
+      output_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * output_sampling_khz_)),
+      data_pointer_(0),
+      loop_length_samples_(0),
+      max_bytes_(0),
+      encoded_bytes_(0),
+      encoding_time_ms_(0.0),
+      decoding_time_ms_(0.0),
+      out_file_(NULL) {}
+
+void AudioCodecSpeedTest::SetUp() {
+  channels_ = get<0>(GetParam());
+  bit_rate_ = get<1>(GetParam());
+  in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
+  save_out_data_ = get<4>(GetParam());
+
+  FILE* fp = fopen(in_filename_.c_str(), "rb");
+  RTC_DCHECK(fp);
+
+  // Obtain file size.
+  fseek(fp, 0, SEEK_END);
+  loop_length_samples_ = ftell(fp) / sizeof(int16_t);
+  rewind(fp);
+
+  // Allocate memory to contain the whole file.
+  in_data_.reset(
+      new int16_t[loop_length_samples_ + input_length_sample_ * channels_]);
+
+  data_pointer_ = 0;
+
+  // Copy the file into the buffer.
+  ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
+            loop_length_samples_);
+  fclose(fp);
+
+  // Add an extra block length of samples to the end of the array, starting
+  // over again from the beginning of the array. This is done to simplify
+  // the reading process when reading over the end of the loop.
+  memcpy(&in_data_[loop_length_samples_], &in_data_[0],
+         input_length_sample_ * channels_ * sizeof(int16_t));
+
+  max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
+  out_data_.reset(new int16_t[output_length_sample_ * channels_]);
+  bit_stream_.reset(new uint8_t[max_bytes_]);
+
+  if (save_out_data_) {
+    std::string out_filename =
+        ::testing::UnitTest::GetInstance()->current_test_info()->name();
+
+    // Erase '/'
+    size_t found;
+    while ((found = out_filename.find('/')) != std::string::npos)
+      out_filename.replace(found, 1, "_");
+
+    out_filename = test::OutputPath() + out_filename + ".pcm";
+
+    out_file_ = fopen(out_filename.c_str(), "wb");
+    RTC_DCHECK(out_file_);
+
+    printf("Output to be saved in %s.\n", out_filename.c_str());
+  }
+}
+
+void AudioCodecSpeedTest::TearDown() {
+  if (save_out_data_) {
+    fclose(out_file_);
+  }
+}
+
+void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
+  size_t time_now_ms = 0;
+  float time_ms;
+
+  printf("Coding %d kHz-sampled %zu-channel audio at %d bps ...\n",
+         input_sampling_khz_, channels_, bit_rate_);
+
+  while (time_now_ms < audio_duration_sec * 1000) {
+    // Encode & decode.
+    time_ms = EncodeABlock(&in_data_[data_pointer_], &bit_stream_[0],
+                           max_bytes_, &encoded_bytes_);
+    encoding_time_ms_ += time_ms;
+    time_ms = DecodeABlock(&bit_stream_[0], encoded_bytes_, &out_data_[0]);
+    decoding_time_ms_ += time_ms;
+    if (save_out_data_) {
+      fwrite(&out_data_[0], sizeof(int16_t), output_length_sample_ * channels_,
+             out_file_);
+    }
+    data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
+                    loop_length_samples_;
+    time_now_ms += block_duration_ms_;
+  }
+
+  printf("Encoding: %.2f%% real time,\nDecoding: %.2f%% real time.\n",
+         (encoding_time_ms_ / audio_duration_sec) / 10.0,
+         (decoding_time_ms_ / audio_duration_sec) / 10.0);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
new file mode 100644
index 0000000000..c5f1d7c259
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+#define MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+
+#include <memory>
+#include <string>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// Define coding parameter as
+// <channels, bit_rate, file_name, extension, if_save_output>.
+typedef std::tuple<size_t, int, std::string, std::string, bool> coding_param;
+
+class AudioCodecSpeedTest : public ::testing::TestWithParam<coding_param> {
+ protected:
+  AudioCodecSpeedTest(int block_duration_ms,
+                      int input_sampling_khz,
+                      int output_sampling_khz);
+  virtual void SetUp();
+  virtual void TearDown();
+
+  // EncodeABlock(...) does the following:
+  // 1. encodes a block of audio, saved in `in_data`,
+  // 2. save the bit stream to `bit_stream` of `max_bytes` bytes in size,
+  // 3. assign `encoded_bytes` with the length of the bit stream (in bytes),
+  // 4. return the cost of time (in millisecond) spent on actual encoding.
+  virtual float EncodeABlock(int16_t* in_data,
+                             uint8_t* bit_stream,
+                             size_t max_bytes,
+                             size_t* encoded_bytes) = 0;
+
+  // DecodeABlock(...) does the following:
+  // 1. decodes the bit stream in `bit_stream` with a length of `encoded_bytes`
+  // (in bytes),
+  // 2. save the decoded audio in `out_data`,
+  // 3. return the cost of time (in millisecond) spent on actual decoding.
+  virtual float DecodeABlock(const uint8_t* bit_stream,
+                             size_t encoded_bytes,
+                             int16_t* out_data) = 0;
+
+  // Encoding and decode an audio of `audio_duration` (in seconds) and
+  // record the runtime for encoding and decoding separately.
+  void EncodeDecode(size_t audio_duration);
+
+  int block_duration_ms_;
+  int input_sampling_khz_;
+  int output_sampling_khz_;
+
+  // Number of samples-per-channel in a frame.
+  size_t input_length_sample_;
+
+  // Expected output number of samples-per-channel in a frame.
+  size_t output_length_sample_;
+
+  std::unique_ptr<int16_t[]> in_data_;
+  std::unique_ptr<int16_t[]> out_data_;
+  size_t data_pointer_;
+  size_t loop_length_samples_;
+  std::unique_ptr<uint8_t[]> bit_stream_;
+
+  // Maximum number of bytes in output bitstream for a frame of audio.
+  size_t max_bytes_;
+
+  size_t encoded_bytes_;
+  float encoding_time_ms_;
+  float decoding_time_ms_;
+  FILE* out_file_;
+
+  size_t channels_;
+
+  // Bit rate is in bit-per-second.
+  int bit_rate_;
+
+  std::string in_filename_;
+
+  // Determines whether to save the output to file.
+  bool save_out_data_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_