Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

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

187 files changed, 32563 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc
new file mode 100644
index 0000000000..f4ef6cdccb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc
@@ -0,0 +1,106 @@
+/*
+ *  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/neteq/accelerate.h"
+
+
+#include "api/array_view.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+namespace webrtc {
+
+Accelerate::ReturnCodes Accelerate::Process(const int16_t* input,
+                                            size_t input_length,
+                                            bool fast_accelerate,
+                                            AudioMultiVector* output,
+                                            size_t* length_change_samples) {
+  // Input length must be (almost) 30 ms.
+  static const size_t k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
+  if (num_channels_ == 0 ||
+      input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) {
+    // Length of input data too short to do accelerate. Simply move all data
+    // from input to output.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, input_length));
+    return kError;
+  }
+  return TimeStretch::Process(input, input_length, fast_accelerate, output,
+                              length_change_samples);
+}
+
+void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/,
+                                               int16_t* best_correlation,
+                                               size_t* /*peak_index*/) const {
+  // When the signal does not contain any active speech, the correlation does
+  // not matter. Simply set it to zero.
+  *best_correlation = 0;
+}
+
+Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch(
+    const int16_t* input,
+    size_t input_length,
+    size_t peak_index,
+    int16_t best_correlation,
+    bool active_speech,
+    bool fast_mode,
+    AudioMultiVector* output) const {
+  // Check for strong correlation or passive speech.
+  // Use 8192 (0.5 in Q14) in fast mode.
+  const int correlation_threshold = fast_mode ? 8192 : kCorrelationThreshold;
+  if ((best_correlation > correlation_threshold) || !active_speech) {
+    // Do accelerate operation by overlap add.
+
+    // Pre-calculate common multiplication with `fs_mult_`.
+    // 120 corresponds to 15 ms.
+    size_t fs_mult_120 = fs_mult_ * 120;
+
+    if (fast_mode) {
+      // Fit as many multiples of `peak_index` as possible in fs_mult_120.
+      // TODO(henrik.lundin) Consider finding multiple correlation peaks and
+      // pick the one with the longest correlation lag in this case.
+      peak_index = (fs_mult_120 / peak_index) * peak_index;
+    }
+
+    RTC_DCHECK_GE(fs_mult_120, peak_index);  // Should be handled in Process().
+    // Copy first part; 0 to 15 ms.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, fs_mult_120 * num_channels_));
+    // Copy the `peak_index` starting at 15 ms to `temp_vector`.
+    AudioMultiVector temp_vector(num_channels_);
+    temp_vector.PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        &input[fs_mult_120 * num_channels_], peak_index * num_channels_));
+    // Cross-fade `temp_vector` onto the end of `output`.
+    output->CrossFade(temp_vector, peak_index);
+    // Copy the last unmodified part, 15 ms + pitch period until the end.
+    output->PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        &input[(fs_mult_120 + peak_index) * num_channels_],
+        input_length - (fs_mult_120 + peak_index) * num_channels_));
+
+    if (active_speech) {
+      return kSuccess;
+    } else {
+      return kSuccessLowEnergy;
+    }
+  } else {
+    // Accelerate not allowed. Simply move all data from decoded to outData.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, input_length));
+    return kNoStretch;
+  }
+}
+
+Accelerate* AccelerateFactory::Create(
+    int sample_rate_hz,
+    size_t num_channels,
+    const BackgroundNoise& background_noise) const {
+  return new Accelerate(sample_rate_hz, num_channels, background_noise);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.h b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.h
new file mode 100644
index 0000000000..01fe874d54
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.h
@@ -0,0 +1,79 @@
+/*
+ *  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_NETEQ_ACCELERATE_H_
+#define MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/neteq/time_stretch.h"
+
+namespace webrtc {
+
+class AudioMultiVector;
+class BackgroundNoise;
+
+// This class implements the Accelerate operation. Most of the work is done
+// in the base class TimeStretch, which is shared with the PreemptiveExpand
+// operation. In the Accelerate class, the operations that are specific to
+// Accelerate are implemented.
+class Accelerate : public TimeStretch {
+ public:
+  Accelerate(int sample_rate_hz,
+             size_t num_channels,
+             const BackgroundNoise& background_noise)
+      : TimeStretch(sample_rate_hz, num_channels, background_noise) {}
+
+  Accelerate(const Accelerate&) = delete;
+  Accelerate& operator=(const Accelerate&) = delete;
+
+  // This method performs the actual Accelerate operation. The samples are
+  // read from `input`, of length `input_length` elements, and are written to
+  // `output`. The number of samples removed through time-stretching is
+  // is provided in the output `length_change_samples`. The method returns
+  // the outcome of the operation as an enumerator value. If `fast_accelerate`
+  // is true, the algorithm will relax the requirements on finding strong
+  // correlations, and may remove multiple pitch periods if possible.
+  ReturnCodes Process(const int16_t* input,
+                      size_t input_length,
+                      bool fast_accelerate,
+                      AudioMultiVector* output,
+                      size_t* length_change_samples);
+
+ protected:
+  // Sets the parameters `best_correlation` and `peak_index` to suitable
+  // values when the signal contains no active speech.
+  void SetParametersForPassiveSpeech(size_t len,
+                                     int16_t* best_correlation,
+                                     size_t* peak_index) const override;
+
+  // Checks the criteria for performing the time-stretching operation and,
+  // if possible, performs the time-stretching.
+  ReturnCodes CheckCriteriaAndStretch(const int16_t* input,
+                                      size_t input_length,
+                                      size_t peak_index,
+                                      int16_t best_correlation,
+                                      bool active_speech,
+                                      bool fast_mode,
+                                      AudioMultiVector* output) const override;
+};
+
+struct AccelerateFactory {
+  AccelerateFactory() {}
+  virtual ~AccelerateFactory() {}
+
+  virtual Accelerate* Create(int sample_rate_hz,
+                             size_t num_channels,
+                             const BackgroundNoise& background_noise) const;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
new file mode 100644
index 0000000000..fef3c3c1e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -0,0 +1,526 @@
+/*
+ *  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 <stdlib.h>
+
+#include <array>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#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/opus/audio_decoder_opus.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "rtc_base/system/arch.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kOverheadBytesPerPacket = 50;
+
+// The absolute difference between the input and output (the first channel) is
+// compared vs `tolerance`. The parameter `delay` is used to correct for codec
+// delays.
+void CompareInputOutput(const std::vector<int16_t>& input,
+                        const std::vector<int16_t>& output,
+                        size_t num_samples,
+                        size_t channels,
+                        int tolerance,
+                        int delay) {
+  ASSERT_LE(num_samples, input.size());
+  ASSERT_LE(num_samples * channels, output.size());
+  for (unsigned int n = 0; n < num_samples - delay; ++n) {
+    ASSERT_NEAR(input[n], output[channels * n + delay], tolerance)
+        << "Exit test on first diff; n = " << n;
+  }
+}
+
+// The absolute difference between the first two channels in `output` is
+// compared vs `tolerance`.
+void CompareTwoChannels(const std::vector<int16_t>& output,
+                        size_t samples_per_channel,
+                        size_t channels,
+                        int tolerance) {
+  ASSERT_GE(channels, 2u);
+  ASSERT_LE(samples_per_channel * channels, output.size());
+  for (unsigned int n = 0; n < samples_per_channel; ++n)
+    ASSERT_NEAR(output[channels * n], output[channels * n + 1], tolerance)
+        << "Stereo samples differ.";
+}
+
+// Calculates mean-squared error between input and output (the first channel).
+// The parameter `delay` is used to correct for codec delays.
+double MseInputOutput(const std::vector<int16_t>& input,
+                      const std::vector<int16_t>& output,
+                      size_t num_samples,
+                      size_t channels,
+                      int delay) {
+  RTC_DCHECK_LT(delay, static_cast<int>(num_samples));
+  RTC_DCHECK_LE(num_samples, input.size());
+  RTC_DCHECK_LE(num_samples * channels, output.size());
+  if (num_samples == 0)
+    return 0.0;
+  double squared_sum = 0.0;
+  for (unsigned int n = 0; n < num_samples - delay; ++n) {
+    squared_sum += (input[n] - output[channels * n + delay]) *
+                   (input[n] - output[channels * n + delay]);
+  }
+  return squared_sum / (num_samples - delay);
+}
+}  // namespace
+
+class AudioDecoderTest : public ::testing::Test {
+ protected:
+  AudioDecoderTest()
+      : input_audio_(
+            webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+            32000),
+        codec_input_rate_hz_(32000),  // Legacy default value.
+        frame_size_(0),
+        data_length_(0),
+        channels_(1),
+        payload_type_(17),
+        decoder_(NULL) {}
+
+  ~AudioDecoderTest() override {}
+
+  void SetUp() override {
+    if (audio_encoder_)
+      codec_input_rate_hz_ = audio_encoder_->SampleRateHz();
+    // Create arrays.
+    ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0";
+  }
+
+  void TearDown() override {
+    delete decoder_;
+    decoder_ = NULL;
+  }
+
+  virtual void InitEncoder() {}
+
+  // TODO(henrik.lundin) Change return type to size_t once most/all overriding
+  // implementations are gone.
+  virtual int EncodeFrame(const int16_t* input,
+                          size_t input_len_samples,
+                          rtc::Buffer* output) {
+    AudioEncoder::EncodedInfo encoded_info;
+    const size_t samples_per_10ms = audio_encoder_->SampleRateHz() / 100;
+    RTC_CHECK_EQ(samples_per_10ms * audio_encoder_->Num10MsFramesInNextPacket(),
+                 input_len_samples);
+    std::unique_ptr<int16_t[]> interleaved_input(
+        new int16_t[channels_ * samples_per_10ms]);
+    for (size_t i = 0; i < audio_encoder_->Num10MsFramesInNextPacket(); ++i) {
+      EXPECT_EQ(0u, encoded_info.encoded_bytes);
+
+      // Duplicate the mono input signal to however many channels the test
+      // wants.
+      test::InputAudioFile::DuplicateInterleaved(input + i * samples_per_10ms,
+                                                 samples_per_10ms, channels_,
+                                                 interleaved_input.get());
+
+      encoded_info =
+          audio_encoder_->Encode(0,
+                                 rtc::ArrayView<const int16_t>(
+                                     interleaved_input.get(),
+                                     audio_encoder_->NumChannels() *
+                                         audio_encoder_->SampleRateHz() / 100),
+                                 output);
+    }
+    EXPECT_EQ(payload_type_, encoded_info.payload_type);
+    return static_cast<int>(encoded_info.encoded_bytes);
+  }
+
+  // Encodes and decodes audio. The absolute difference between the input and
+  // output is compared vs `tolerance`, and the mean-squared error is compared
+  // with `mse`. The encoded stream should contain `expected_bytes`. For stereo
+  // audio, the absolute difference between the two channels is compared vs
+  // `channel_diff_tolerance`.
+  void EncodeDecodeTest(size_t expected_bytes,
+                        int tolerance,
+                        double mse,
+                        int delay = 0,
+                        int channel_diff_tolerance = 0) {
+    ASSERT_GE(tolerance, 0) << "Test must define a tolerance >= 0";
+    ASSERT_GE(channel_diff_tolerance, 0)
+        << "Test must define a channel_diff_tolerance >= 0";
+    size_t processed_samples = 0u;
+    size_t encoded_bytes = 0u;
+    InitEncoder();
+    std::vector<int16_t> input;
+    std::vector<int16_t> decoded;
+    while (processed_samples + frame_size_ <= data_length_) {
+      // Extend input vector with `frame_size_`.
+      input.resize(input.size() + frame_size_, 0);
+      // Read from input file.
+      ASSERT_GE(input.size() - processed_samples, frame_size_);
+      ASSERT_TRUE(input_audio_.Read(frame_size_, codec_input_rate_hz_,
+                                    &input[processed_samples]));
+      rtc::Buffer encoded;
+      size_t enc_len =
+          EncodeFrame(&input[processed_samples], frame_size_, &encoded);
+      // Make sure that frame_size_ * channels_ samples are allocated and free.
+      decoded.resize((processed_samples + frame_size_) * channels_, 0);
+
+      const std::vector<AudioDecoder::ParseResult> parse_result =
+          decoder_->ParsePayload(std::move(encoded), /*timestamp=*/0);
+      RTC_CHECK_EQ(parse_result.size(), size_t{1});
+      auto decode_result = parse_result[0].frame->Decode(
+          rtc::ArrayView<int16_t>(&decoded[processed_samples * channels_],
+                                  frame_size_ * channels_ * sizeof(int16_t)));
+      RTC_CHECK(decode_result.has_value());
+      EXPECT_EQ(frame_size_ * channels_, decode_result->num_decoded_samples);
+      encoded_bytes += enc_len;
+      processed_samples += frame_size_;
+    }
+    // For some codecs it doesn't make sense to check expected number of bytes,
+    // since the number can vary for different platforms. Opus is such a codec.
+    // In this case expected_bytes is set to 0.
+    if (expected_bytes) {
+      EXPECT_EQ(expected_bytes, encoded_bytes);
+    }
+    CompareInputOutput(input, decoded, processed_samples, channels_, tolerance,
+                       delay);
+    if (channels_ == 2)
+      CompareTwoChannels(decoded, processed_samples, channels_,
+                         channel_diff_tolerance);
+    EXPECT_LE(
+        MseInputOutput(input, decoded, processed_samples, channels_, delay),
+        mse);
+  }
+
+  // Encodes a payload and decodes it twice with decoder re-init before each
+  // decode. Verifies that the decoded result is the same.
+  void ReInitTest() {
+    InitEncoder();
+    std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);
+    ASSERT_TRUE(
+        input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
+    std::array<rtc::Buffer, 2> encoded;
+    EncodeFrame(input.get(), frame_size_, &encoded[0]);
+    // Make a copy.
+    encoded[1].SetData(encoded[0].data(), encoded[0].size());
+
+    std::array<std::vector<int16_t>, 2> outputs;
+    for (size_t i = 0; i < outputs.size(); ++i) {
+      outputs[i].resize(frame_size_ * channels_);
+      decoder_->Reset();
+      const std::vector<AudioDecoder::ParseResult> parse_result =
+          decoder_->ParsePayload(std::move(encoded[i]), /*timestamp=*/0);
+      RTC_CHECK_EQ(parse_result.size(), size_t{1});
+      auto decode_result = parse_result[0].frame->Decode(outputs[i]);
+      RTC_CHECK(decode_result.has_value());
+      EXPECT_EQ(frame_size_ * channels_, decode_result->num_decoded_samples);
+    }
+    EXPECT_EQ(outputs[0], outputs[1]);
+  }
+
+  // Call DecodePlc and verify that the correct number of samples is produced.
+  void DecodePlcTest() {
+    InitEncoder();
+    std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);
+    ASSERT_TRUE(
+        input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
+    rtc::Buffer encoded;
+    EncodeFrame(input.get(), frame_size_, &encoded);
+    decoder_->Reset();
+    std::vector<int16_t> output(frame_size_ * channels_);
+    const std::vector<AudioDecoder::ParseResult> parse_result =
+        decoder_->ParsePayload(std::move(encoded), /*timestamp=*/0);
+    RTC_CHECK_EQ(parse_result.size(), size_t{1});
+    auto decode_result = parse_result[0].frame->Decode(output);
+    RTC_CHECK(decode_result.has_value());
+    EXPECT_EQ(frame_size_ * channels_, decode_result->num_decoded_samples);
+    // Call DecodePlc and verify that we get one frame of data.
+    // (Overwrite the output from the above Decode call, but that does not
+    // matter.)
+    size_t dec_len =
+        decoder_->DecodePlc(/*num_frames=*/1, /*decoded=*/output.data());
+    EXPECT_EQ(frame_size_ * channels_, dec_len);
+  }
+
+  test::ResampleInputAudioFile input_audio_;
+  int codec_input_rate_hz_;
+  size_t frame_size_;
+  size_t data_length_;
+  size_t channels_;
+  const int payload_type_;
+  AudioDecoder* decoder_;
+  std::unique_ptr<AudioEncoder> audio_encoder_;
+};
+
+class AudioDecoderPcmUTest : public AudioDecoderTest {
+ protected:
+  AudioDecoderPcmUTest() : AudioDecoderTest() {
+    frame_size_ = 160;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderPcmU(1);
+    AudioEncoderPcmU::Config config;
+    config.frame_size_ms = static_cast<int>(frame_size_ / 8);
+    config.payload_type = payload_type_;
+    audio_encoder_.reset(new AudioEncoderPcmU(config));
+  }
+};
+
+class AudioDecoderPcmATest : public AudioDecoderTest {
+ protected:
+  AudioDecoderPcmATest() : AudioDecoderTest() {
+    frame_size_ = 160;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderPcmA(1);
+    AudioEncoderPcmA::Config config;
+    config.frame_size_ms = static_cast<int>(frame_size_ / 8);
+    config.payload_type = payload_type_;
+    audio_encoder_.reset(new AudioEncoderPcmA(config));
+  }
+};
+
+class AudioDecoderPcm16BTest : public AudioDecoderTest {
+ protected:
+  AudioDecoderPcm16BTest() : AudioDecoderTest() {
+    codec_input_rate_hz_ = 16000;
+    frame_size_ = 20 * codec_input_rate_hz_ / 1000;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderPcm16B(codec_input_rate_hz_, 1);
+    RTC_DCHECK(decoder_);
+    AudioEncoderPcm16B::Config config;
+    config.sample_rate_hz = codec_input_rate_hz_;
+    config.frame_size_ms =
+        static_cast<int>(frame_size_ / (config.sample_rate_hz / 1000));
+    config.payload_type = payload_type_;
+    audio_encoder_.reset(new AudioEncoderPcm16B(config));
+  }
+};
+
+class AudioDecoderIlbcTest : public AudioDecoderTest {
+ protected:
+  AudioDecoderIlbcTest() : AudioDecoderTest() {
+    codec_input_rate_hz_ = 8000;
+    frame_size_ = 240;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderIlbcImpl;
+    RTC_DCHECK(decoder_);
+    AudioEncoderIlbcConfig config;
+    config.frame_size_ms = 30;
+    audio_encoder_.reset(new AudioEncoderIlbcImpl(config, payload_type_));
+  }
+
+  // Overload the default test since iLBC's function WebRtcIlbcfix_NetEqPlc does
+  // not return any data. It simply resets a few states and returns 0.
+  void DecodePlcTest() {
+    InitEncoder();
+    std::unique_ptr<int16_t[]> input(new int16_t[frame_size_]);
+    ASSERT_TRUE(
+        input_audio_.Read(frame_size_, codec_input_rate_hz_, input.get()));
+    rtc::Buffer encoded;
+    size_t enc_len = EncodeFrame(input.get(), frame_size_, &encoded);
+    AudioDecoder::SpeechType speech_type;
+    decoder_->Reset();
+    std::unique_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
+    size_t dec_len = decoder_->Decode(
+        encoded.data(), enc_len, codec_input_rate_hz_,
+        frame_size_ * channels_ * sizeof(int16_t), output.get(), &speech_type);
+    EXPECT_EQ(frame_size_, dec_len);
+    // Simply call DecodePlc and verify that we get 0 as return value.
+    EXPECT_EQ(0U, decoder_->DecodePlc(1, output.get()));
+  }
+};
+
+class AudioDecoderG722Test : public AudioDecoderTest {
+ protected:
+  AudioDecoderG722Test() : AudioDecoderTest() {
+    codec_input_rate_hz_ = 16000;
+    frame_size_ = 160;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderG722Impl;
+    RTC_DCHECK(decoder_);
+    AudioEncoderG722Config config;
+    config.frame_size_ms = 10;
+    config.num_channels = 1;
+    audio_encoder_.reset(new AudioEncoderG722Impl(config, payload_type_));
+  }
+};
+
+class AudioDecoderG722StereoTest : public AudioDecoderTest {
+ protected:
+  AudioDecoderG722StereoTest() : AudioDecoderTest() {
+    channels_ = 2;
+    codec_input_rate_hz_ = 16000;
+    frame_size_ = 160;
+    data_length_ = 10 * frame_size_;
+    decoder_ = new AudioDecoderG722StereoImpl;
+    RTC_DCHECK(decoder_);
+    AudioEncoderG722Config config;
+    config.frame_size_ms = 10;
+    config.num_channels = 2;
+    audio_encoder_.reset(new AudioEncoderG722Impl(config, payload_type_));
+  }
+};
+
+class AudioDecoderOpusTest
+    : public AudioDecoderTest,
+      public testing::WithParamInterface<std::tuple<int, int>> {
+ protected:
+  AudioDecoderOpusTest() : AudioDecoderTest() {
+    channels_ = opus_num_channels_;
+    codec_input_rate_hz_ = opus_sample_rate_hz_;
+    frame_size_ = rtc::CheckedDivExact(opus_sample_rate_hz_, 100);
+    data_length_ = 10 * frame_size_;
+    decoder_ =
+        new AudioDecoderOpusImpl(opus_num_channels_, opus_sample_rate_hz_);
+    AudioEncoderOpusConfig config;
+    config.frame_size_ms = 10;
+    config.sample_rate_hz = opus_sample_rate_hz_;
+    config.num_channels = opus_num_channels_;
+    config.application = opus_num_channels_ == 1
+                             ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                             : AudioEncoderOpusConfig::ApplicationMode::kAudio;
+    audio_encoder_ = AudioEncoderOpus::MakeAudioEncoder(config, payload_type_);
+    audio_encoder_->OnReceivedOverhead(kOverheadBytesPerPacket);
+  }
+  const int opus_sample_rate_hz_{std::get<0>(GetParam())};
+  const int opus_num_channels_{std::get<1>(GetParam())};
+};
+
+INSTANTIATE_TEST_SUITE_P(Param,
+                         AudioDecoderOpusTest,
+                         testing::Combine(testing::Values(16000, 48000),
+                                          testing::Values(1, 2)));
+
+TEST_F(AudioDecoderPcmUTest, EncodeDecode) {
+  int tolerance = 251;
+  double mse = 1734.0;
+  EncodeDecodeTest(data_length_, tolerance, mse);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+namespace {
+int SetAndGetTargetBitrate(AudioEncoder* audio_encoder, int rate) {
+  audio_encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
+  return audio_encoder->GetTargetBitrate();
+}
+void TestSetAndGetTargetBitratesWithFixedCodec(AudioEncoder* audio_encoder,
+                                               int fixed_rate) {
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, 32000));
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate - 1));
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate));
+  EXPECT_EQ(fixed_rate, SetAndGetTargetBitrate(audio_encoder, fixed_rate + 1));
+}
+}  // namespace
+
+TEST_F(AudioDecoderPcmUTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
+}
+
+TEST_F(AudioDecoderPcmATest, EncodeDecode) {
+  int tolerance = 308;
+  double mse = 1931.0;
+  EncodeDecodeTest(data_length_, tolerance, mse);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+TEST_F(AudioDecoderPcmATest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
+}
+
+TEST_F(AudioDecoderPcm16BTest, EncodeDecode) {
+  int tolerance = 0;
+  double mse = 0.0;
+  EncodeDecodeTest(2 * data_length_, tolerance, mse);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+TEST_F(AudioDecoderPcm16BTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(),
+                                            codec_input_rate_hz_ * 16);
+}
+
+TEST_F(AudioDecoderIlbcTest, EncodeDecode) {
+  int tolerance = 6808;
+  double mse = 2.13e6;
+  int delay = 80;  // Delay from input to output.
+  EncodeDecodeTest(500, tolerance, mse, delay);
+  ReInitTest();
+  EXPECT_TRUE(decoder_->HasDecodePlc());
+  DecodePlcTest();
+}
+
+TEST_F(AudioDecoderIlbcTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 13333);
+}
+
+TEST_F(AudioDecoderG722Test, EncodeDecode) {
+  int tolerance = 6176;
+  double mse = 238630.0;
+  int delay = 22;  // Delay from input to output.
+  EncodeDecodeTest(data_length_ / 2, tolerance, mse, delay);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+TEST_F(AudioDecoderG722Test, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 64000);
+}
+
+TEST_F(AudioDecoderG722StereoTest, EncodeDecode) {
+  int tolerance = 6176;
+  int channel_diff_tolerance = 0;
+  double mse = 238630.0;
+  int delay = 22;  // Delay from input to output.
+  EncodeDecodeTest(data_length_, tolerance, mse, delay, channel_diff_tolerance);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+TEST_F(AudioDecoderG722StereoTest, SetTargetBitrate) {
+  TestSetAndGetTargetBitratesWithFixedCodec(audio_encoder_.get(), 128000);
+}
+
+// TODO(http://bugs.webrtc.org/12518): Enable the test after Opus has been
+// updated.
+TEST_P(AudioDecoderOpusTest, DISABLED_EncodeDecode) {
+  constexpr int tolerance = 6176;
+  constexpr int channel_diff_tolerance = 6;
+  constexpr double mse = 238630.0;
+  constexpr int delay = 22;  // Delay from input to output.
+  EncodeDecodeTest(0, tolerance, mse, delay, channel_diff_tolerance);
+  ReInitTest();
+  EXPECT_FALSE(decoder_->HasDecodePlc());
+}
+
+TEST_P(AudioDecoderOpusTest, SetTargetBitrate) {
+  const int overhead_rate =
+      8 * kOverheadBytesPerPacket * codec_input_rate_hz_ / frame_size_;
+  EXPECT_EQ(6000,
+            SetAndGetTargetBitrate(audio_encoder_.get(), 5999 + overhead_rate));
+  EXPECT_EQ(6000,
+            SetAndGetTargetBitrate(audio_encoder_.get(), 6000 + overhead_rate));
+  EXPECT_EQ(32000, SetAndGetTargetBitrate(audio_encoder_.get(),
+                                          32000 + overhead_rate));
+  EXPECT_EQ(510000, SetAndGetTargetBitrate(audio_encoder_.get(),
+                                           510000 + overhead_rate));
+  EXPECT_EQ(510000, SetAndGetTargetBitrate(audio_encoder_.get(),
+                                           511000 + overhead_rate));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.cc b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.cc
new file mode 100644
index 0000000000..14ae94649b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.cc
@@ -0,0 +1,222 @@
+/*
+ *  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/neteq/audio_multi_vector.h"
+
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioMultiVector::AudioMultiVector(size_t N) {
+  RTC_DCHECK_GT(N, 0);
+  if (N < 1)
+    N = 1;
+  for (size_t n = 0; n < N; ++n) {
+    channels_.push_back(new AudioVector);
+  }
+  num_channels_ = N;
+}
+
+AudioMultiVector::AudioMultiVector(size_t N, size_t initial_size) {
+  RTC_DCHECK_GT(N, 0);
+  if (N < 1)
+    N = 1;
+  for (size_t n = 0; n < N; ++n) {
+    channels_.push_back(new AudioVector(initial_size));
+  }
+  num_channels_ = N;
+}
+
+AudioMultiVector::~AudioMultiVector() {
+  std::vector<AudioVector*>::iterator it = channels_.begin();
+  while (it != channels_.end()) {
+    delete (*it);
+    ++it;
+  }
+}
+
+void AudioMultiVector::Clear() {
+  for (size_t i = 0; i < num_channels_; ++i) {
+    channels_[i]->Clear();
+  }
+}
+
+void AudioMultiVector::Zeros(size_t length) {
+  for (size_t i = 0; i < num_channels_; ++i) {
+    channels_[i]->Clear();
+    channels_[i]->Extend(length);
+  }
+}
+
+void AudioMultiVector::CopyTo(AudioMultiVector* copy_to) const {
+  if (copy_to) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      channels_[i]->CopyTo(&(*copy_to)[i]);
+    }
+  }
+}
+
+void AudioMultiVector::PushBackInterleaved(
+    rtc::ArrayView<const int16_t> append_this) {
+  RTC_DCHECK_EQ(append_this.size() % num_channels_, 0);
+  if (append_this.empty()) {
+    return;
+  }
+  if (num_channels_ == 1) {
+    // Special case to avoid extra allocation and data shuffling.
+    channels_[0]->PushBack(append_this.data(), append_this.size());
+    return;
+  }
+  size_t length_per_channel = append_this.size() / num_channels_;
+  int16_t* temp_array = new int16_t[length_per_channel];  // Temporary storage.
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    // Copy elements to `temp_array`.
+    for (size_t i = 0; i < length_per_channel; ++i) {
+      temp_array[i] = append_this[channel + i * num_channels_];
+    }
+    channels_[channel]->PushBack(temp_array, length_per_channel);
+  }
+  delete[] temp_array;
+}
+
+void AudioMultiVector::PushBack(const AudioMultiVector& append_this) {
+  RTC_DCHECK_EQ(num_channels_, append_this.num_channels_);
+  if (num_channels_ == append_this.num_channels_) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      channels_[i]->PushBack(append_this[i]);
+    }
+  }
+}
+
+void AudioMultiVector::PushBackFromIndex(const AudioMultiVector& append_this,
+                                         size_t index) {
+  RTC_DCHECK_LT(index, append_this.Size());
+  index = std::min(index, append_this.Size() - 1);
+  size_t length = append_this.Size() - index;
+  RTC_DCHECK_EQ(num_channels_, append_this.num_channels_);
+  if (num_channels_ == append_this.num_channels_) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      channels_[i]->PushBack(append_this[i], length, index);
+    }
+  }
+}
+
+void AudioMultiVector::PopFront(size_t length) {
+  for (size_t i = 0; i < num_channels_; ++i) {
+    channels_[i]->PopFront(length);
+  }
+}
+
+void AudioMultiVector::PopBack(size_t length) {
+  for (size_t i = 0; i < num_channels_; ++i) {
+    channels_[i]->PopBack(length);
+  }
+}
+
+size_t AudioMultiVector::ReadInterleaved(size_t length,
+                                         int16_t* destination) const {
+  return ReadInterleavedFromIndex(0, length, destination);
+}
+
+size_t AudioMultiVector::ReadInterleavedFromIndex(size_t start_index,
+                                                  size_t length,
+                                                  int16_t* destination) const {
+  RTC_DCHECK(destination);
+  size_t index = 0;  // Number of elements written to `destination` so far.
+  RTC_DCHECK_LE(start_index, Size());
+  start_index = std::min(start_index, Size());
+  if (length + start_index > Size()) {
+    length = Size() - start_index;
+  }
+  if (num_channels_ == 1) {
+    // Special case to avoid the nested for loop below.
+    (*this)[0].CopyTo(length, start_index, destination);
+    return length;
+  }
+  for (size_t i = 0; i < length; ++i) {
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      destination[index] = (*this)[channel][i + start_index];
+      ++index;
+    }
+  }
+  return index;
+}
+
+size_t AudioMultiVector::ReadInterleavedFromEnd(size_t length,
+                                                int16_t* destination) const {
+  length = std::min(length, Size());  // Cannot read more than Size() elements.
+  return ReadInterleavedFromIndex(Size() - length, length, destination);
+}
+
+void AudioMultiVector::OverwriteAt(const AudioMultiVector& insert_this,
+                                   size_t length,
+                                   size_t position) {
+  RTC_DCHECK_EQ(num_channels_, insert_this.num_channels_);
+  // Cap `length` at the length of `insert_this`.
+  RTC_DCHECK_LE(length, insert_this.Size());
+  length = std::min(length, insert_this.Size());
+  if (num_channels_ == insert_this.num_channels_) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      channels_[i]->OverwriteAt(insert_this[i], length, position);
+    }
+  }
+}
+
+void AudioMultiVector::CrossFade(const AudioMultiVector& append_this,
+                                 size_t fade_length) {
+  RTC_DCHECK_EQ(num_channels_, append_this.num_channels_);
+  if (num_channels_ == append_this.num_channels_) {
+    for (size_t i = 0; i < num_channels_; ++i) {
+      channels_[i]->CrossFade(append_this[i], fade_length);
+    }
+  }
+}
+
+size_t AudioMultiVector::Channels() const {
+  return num_channels_;
+}
+
+size_t AudioMultiVector::Size() const {
+  RTC_DCHECK(channels_[0]);
+  return channels_[0]->Size();
+}
+
+void AudioMultiVector::AssertSize(size_t required_size) {
+  if (Size() < required_size) {
+    size_t extend_length = required_size - Size();
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      channels_[channel]->Extend(extend_length);
+    }
+  }
+}
+
+bool AudioMultiVector::Empty() const {
+  RTC_DCHECK(channels_[0]);
+  return channels_[0]->Empty();
+}
+
+void AudioMultiVector::CopyChannel(size_t from_channel, size_t to_channel) {
+  RTC_DCHECK_LT(from_channel, num_channels_);
+  RTC_DCHECK_LT(to_channel, num_channels_);
+  channels_[from_channel]->CopyTo(channels_[to_channel]);
+}
+
+const AudioVector& AudioMultiVector::operator[](size_t index) const {
+  return *(channels_[index]);
+}
+
+AudioVector& AudioMultiVector::operator[](size_t index) {
+  return *(channels_[index]);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.h b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.h
new file mode 100644
index 0000000000..715ec6dfc7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector.h
@@ -0,0 +1,138 @@
+/*
+ *  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_NETEQ_AUDIO_MULTI_VECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
+
+#include <stdint.h>
+#include <string.h>
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
+
+namespace webrtc {
+
+class AudioMultiVector {
+ public:
+  // Creates an empty AudioMultiVector with `N` audio channels. `N` must be
+  // larger than 0.
+  explicit AudioMultiVector(size_t N);
+
+  // Creates an AudioMultiVector with `N` audio channels, each channel having
+  // an initial size. `N` must be larger than 0.
+  AudioMultiVector(size_t N, size_t initial_size);
+
+  virtual ~AudioMultiVector();
+
+  AudioMultiVector(const AudioMultiVector&) = delete;
+  AudioMultiVector& operator=(const AudioMultiVector&) = delete;
+
+  // Deletes all values and make the vector empty.
+  virtual void Clear();
+
+  // Clears the vector and inserts `length` zeros into each channel.
+  virtual void Zeros(size_t length);
+
+  // Copies all values from this vector to `copy_to`. Any contents in `copy_to`
+  // are deleted. After the operation is done, `copy_to` will be an exact
+  // replica of this object. The source and the destination must have the same
+  // number of channels.
+  virtual void CopyTo(AudioMultiVector* copy_to) const;
+
+  // Appends the contents of `append_this` to the end of this object. The array
+  // is assumed to be channel-interleaved. The length must be an even multiple
+  // of this object's number of channels. The length of this object is increased
+  // with the length of the array divided by the number of channels.
+  void PushBackInterleaved(rtc::ArrayView<const int16_t> append_this);
+
+  // Appends the contents of AudioMultiVector `append_this` to this object. The
+  // length of this object is increased with the length of `append_this`.
+  virtual void PushBack(const AudioMultiVector& append_this);
+
+  // Appends the contents of AudioMultiVector `append_this` to this object,
+  // taken from `index` up until the end of `append_this`. The length of this
+  // object is increased.
+  virtual void PushBackFromIndex(const AudioMultiVector& append_this,
+                                 size_t index);
+
+  // Removes `length` elements from the beginning of this object, from each
+  // channel.
+  virtual void PopFront(size_t length);
+
+  // Removes `length` elements from the end of this object, from each
+  // channel.
+  virtual void PopBack(size_t length);
+
+  // Reads `length` samples from each channel and writes them interleaved to
+  // `destination`. The total number of elements written to `destination` is
+  // returned, i.e., `length` * number of channels. If the AudioMultiVector
+  // contains less than `length` samples per channel, this is reflected in the
+  // return value.
+  virtual size_t ReadInterleaved(size_t length, int16_t* destination) const;
+
+  // Like ReadInterleaved() above, but reads from `start_index` instead of from
+  // the beginning.
+  virtual size_t ReadInterleavedFromIndex(size_t start_index,
+                                          size_t length,
+                                          int16_t* destination) const;
+
+  // Like ReadInterleaved() above, but reads from the end instead of from
+  // the beginning.
+  virtual size_t ReadInterleavedFromEnd(size_t length,
+                                        int16_t* destination) const;
+
+  // Overwrites each channel in this AudioMultiVector with values taken from
+  // `insert_this`. The values are taken from the beginning of `insert_this` and
+  // are inserted starting at `position`. `length` values are written into each
+  // channel. If `length` and `position` are selected such that the new data
+  // extends beyond the end of the current AudioVector, the vector is extended
+  // to accommodate the new data. `length` is limited to the length of
+  // `insert_this`.
+  virtual void OverwriteAt(const AudioMultiVector& insert_this,
+                           size_t length,
+                           size_t position);
+
+  // Appends `append_this` to the end of the current vector. Lets the two
+  // vectors overlap by `fade_length` samples (per channel), and cross-fade
+  // linearly in this region.
+  virtual void CrossFade(const AudioMultiVector& append_this,
+                         size_t fade_length);
+
+  // Returns the number of channels.
+  virtual size_t Channels() const;
+
+  // Returns the number of elements per channel in this AudioMultiVector.
+  virtual size_t Size() const;
+
+  // Verify that each channel can hold at least `required_size` elements. If
+  // not, extend accordingly.
+  virtual void AssertSize(size_t required_size);
+
+  virtual bool Empty() const;
+
+  // Copies the data between two channels in the AudioMultiVector. The method
+  // does not add any new channel. Thus, `from_channel` and `to_channel` must
+  // both be valid channel numbers.
+  virtual void CopyChannel(size_t from_channel, size_t to_channel);
+
+  // Accesses and modifies a channel (i.e., an AudioVector object) of this
+  // AudioMultiVector.
+  const AudioVector& operator[](size_t index) const;
+  AudioVector& operator[](size_t index);
+
+ protected:
+  std::vector<AudioVector*> channels_;
+  size_t num_channels_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
new file mode 100644
index 0000000000..386c3d48a3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
@@ -0,0 +1,323 @@
+/*
+ *  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/neteq/audio_multi_vector.h"
+
+#include <stdlib.h>
+
+#include <string>
+#include <vector>
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+// This is a value-parameterized test. The test cases are instantiated with
+// different values for the test parameter, which is used to determine the
+// number of channels in the AudioMultiBuffer. Note that it is not possible
+// to combine typed testing with value-parameterized testing, and since the
+// tests for AudioVector already covers a number of different type parameters,
+// this test focuses on testing different number of channels, and keeping the
+// value type constant.
+
+class AudioMultiVectorTest : public ::testing::TestWithParam<size_t> {
+ protected:
+  AudioMultiVectorTest()
+      : num_channels_(GetParam()),  // Get the test parameter.
+        array_interleaved_(num_channels_ * array_length()) {}
+
+  ~AudioMultiVectorTest() = default;
+
+  virtual void SetUp() {
+    // Populate test arrays.
+    for (size_t i = 0; i < array_length(); ++i) {
+      array_[i] = static_cast<int16_t>(i);
+    }
+    int16_t* ptr = array_interleaved_.data();
+    // Write 100, 101, 102, ... for first channel.
+    // Write 200, 201, 202, ... for second channel.
+    // And so on.
+    for (size_t i = 0; i < array_length(); ++i) {
+      for (size_t j = 1; j <= num_channels_; ++j) {
+        *ptr = rtc::checked_cast<int16_t>(j * 100 + i);
+        ++ptr;
+      }
+    }
+  }
+
+  size_t array_length() const { return sizeof(array_) / sizeof(array_[0]); }
+
+  const size_t num_channels_;
+  int16_t array_[10];
+  std::vector<int16_t> array_interleaved_;
+};
+
+// Create and destroy AudioMultiVector objects, both empty and with a predefined
+// length.
+TEST_P(AudioMultiVectorTest, CreateAndDestroy) {
+  AudioMultiVector vec1(num_channels_);
+  EXPECT_TRUE(vec1.Empty());
+  EXPECT_EQ(num_channels_, vec1.Channels());
+  EXPECT_EQ(0u, vec1.Size());
+
+  size_t initial_size = 17;
+  AudioMultiVector vec2(num_channels_, initial_size);
+  EXPECT_FALSE(vec2.Empty());
+  EXPECT_EQ(num_channels_, vec2.Channels());
+  EXPECT_EQ(initial_size, vec2.Size());
+}
+
+// Test the subscript operator [] for getting and setting.
+TEST_P(AudioMultiVectorTest, SubscriptOperator) {
+  AudioMultiVector vec(num_channels_, array_length());
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    for (size_t i = 0; i < array_length(); ++i) {
+      vec[channel][i] = static_cast<int16_t>(i);
+      // Make sure to use the const version.
+      const AudioVector& audio_vec = vec[channel];
+      EXPECT_EQ(static_cast<int16_t>(i), audio_vec[i]);
+    }
+  }
+}
+
+// Test the PushBackInterleaved method and the CopyFrom method. The Clear
+// method is also invoked.
+TEST_P(AudioMultiVectorTest, PushBackInterleavedAndCopy) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  AudioMultiVector vec_copy(num_channels_);
+  vec.CopyTo(&vec_copy);  // Copy from `vec` to `vec_copy`.
+  ASSERT_EQ(num_channels_, vec.Channels());
+  ASSERT_EQ(array_length(), vec.Size());
+  ASSERT_EQ(num_channels_, vec_copy.Channels());
+  ASSERT_EQ(array_length(), vec_copy.Size());
+  for (size_t channel = 0; channel < vec.Channels(); ++channel) {
+    for (size_t i = 0; i < array_length(); ++i) {
+      EXPECT_EQ(static_cast<int16_t>((channel + 1) * 100 + i), vec[channel][i]);
+      EXPECT_EQ(vec[channel][i], vec_copy[channel][i]);
+    }
+  }
+
+  // Clear `vec` and verify that it is empty.
+  vec.Clear();
+  EXPECT_TRUE(vec.Empty());
+
+  // Now copy the empty vector and verify that the copy becomes empty too.
+  vec.CopyTo(&vec_copy);
+  EXPECT_TRUE(vec_copy.Empty());
+}
+
+// Try to copy to a NULL pointer. Nothing should happen.
+TEST_P(AudioMultiVectorTest, CopyToNull) {
+  AudioMultiVector vec(num_channels_);
+  AudioMultiVector* vec_copy = NULL;
+  vec.PushBackInterleaved(array_interleaved_);
+  vec.CopyTo(vec_copy);
+}
+
+// Test the PushBack method with another AudioMultiVector as input argument.
+TEST_P(AudioMultiVectorTest, PushBackVector) {
+  AudioMultiVector vec1(num_channels_, array_length());
+  AudioMultiVector vec2(num_channels_, array_length());
+  // Set the first vector to [0, 1, ..., array_length() - 1] +
+  //   100 * channel_number.
+  // Set the second vector to [array_length(), array_length() + 1, ...,
+  //   2 * array_length() - 1] + 100 * channel_number.
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    for (size_t i = 0; i < array_length(); ++i) {
+      vec1[channel][i] = static_cast<int16_t>(i + 100 * channel);
+      vec2[channel][i] =
+          static_cast<int16_t>(i + 100 * channel + array_length());
+    }
+  }
+  // Append vec2 to the back of vec1.
+  vec1.PushBack(vec2);
+  ASSERT_EQ(2u * array_length(), vec1.Size());
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    for (size_t i = 0; i < 2 * array_length(); ++i) {
+      EXPECT_EQ(static_cast<int16_t>(i + 100 * channel), vec1[channel][i]);
+    }
+  }
+}
+
+// Test the PushBackFromIndex method.
+TEST_P(AudioMultiVectorTest, PushBackFromIndex) {
+  AudioMultiVector vec1(num_channels_);
+  vec1.PushBackInterleaved(array_interleaved_);
+  AudioMultiVector vec2(num_channels_);
+
+  // Append vec1 to the back of vec2 (which is empty). Read vec1 from the second
+  // last element.
+  vec2.PushBackFromIndex(vec1, array_length() - 2);
+  ASSERT_EQ(2u, vec2.Size());
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    for (size_t i = 0; i < 2; ++i) {
+      EXPECT_EQ(array_interleaved_[channel +
+                                   num_channels_ * (array_length() - 2 + i)],
+                vec2[channel][i]);
+    }
+  }
+}
+
+// Starts with pushing some values to the vector, then test the Zeros method.
+TEST_P(AudioMultiVectorTest, Zeros) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  vec.Zeros(2 * array_length());
+  ASSERT_EQ(num_channels_, vec.Channels());
+  ASSERT_EQ(2u * array_length(), vec.Size());
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    for (size_t i = 0; i < 2 * array_length(); ++i) {
+      EXPECT_EQ(0, vec[channel][i]);
+    }
+  }
+}
+
+// Test the ReadInterleaved method
+TEST_P(AudioMultiVectorTest, ReadInterleaved) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  int16_t* output = new int16_t[array_interleaved_.size()];
+  // Read 5 samples.
+  size_t read_samples = 5;
+  EXPECT_EQ(num_channels_ * read_samples,
+            vec.ReadInterleaved(read_samples, output));
+  EXPECT_EQ(0, memcmp(array_interleaved_.data(), output,
+                      read_samples * sizeof(int16_t)));
+
+  // Read too many samples. Expect to get all samples from the vector.
+  EXPECT_EQ(array_interleaved_.size(),
+            vec.ReadInterleaved(array_length() + 1, output));
+  EXPECT_EQ(0, memcmp(array_interleaved_.data(), output,
+                      read_samples * sizeof(int16_t)));
+
+  delete[] output;
+}
+
+// Test the PopFront method.
+TEST_P(AudioMultiVectorTest, PopFront) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  vec.PopFront(1);  // Remove one element from each channel.
+  ASSERT_EQ(array_length() - 1u, vec.Size());
+  // Let `ptr` point to the second element of the first channel in the
+  // interleaved array.
+  int16_t* ptr = &array_interleaved_[num_channels_];
+  for (size_t i = 0; i < array_length() - 1; ++i) {
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      EXPECT_EQ(*ptr, vec[channel][i]);
+      ++ptr;
+    }
+  }
+  vec.PopFront(array_length());  // Remove more elements than vector size.
+  EXPECT_EQ(0u, vec.Size());
+}
+
+// Test the PopBack method.
+TEST_P(AudioMultiVectorTest, PopBack) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  vec.PopBack(1);  // Remove one element from each channel.
+  ASSERT_EQ(array_length() - 1u, vec.Size());
+  // Let `ptr` point to the first element of the first channel in the
+  // interleaved array.
+  int16_t* ptr = array_interleaved_.data();
+  for (size_t i = 0; i < array_length() - 1; ++i) {
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      EXPECT_EQ(*ptr, vec[channel][i]);
+      ++ptr;
+    }
+  }
+  vec.PopBack(array_length());  // Remove more elements than vector size.
+  EXPECT_EQ(0u, vec.Size());
+}
+
+// Test the AssertSize method.
+TEST_P(AudioMultiVectorTest, AssertSize) {
+  AudioMultiVector vec(num_channels_, array_length());
+  EXPECT_EQ(array_length(), vec.Size());
+  // Start with asserting with smaller sizes than already allocated.
+  vec.AssertSize(0);
+  vec.AssertSize(array_length() - 1);
+  // Nothing should have changed.
+  EXPECT_EQ(array_length(), vec.Size());
+  // Assert with one element longer than already allocated.
+  vec.AssertSize(array_length() + 1);
+  // Expect vector to have grown.
+  EXPECT_EQ(array_length() + 1, vec.Size());
+  // Also check the individual AudioVectors.
+  for (size_t channel = 0; channel < vec.Channels(); ++channel) {
+    EXPECT_EQ(array_length() + 1u, vec[channel].Size());
+  }
+}
+
+// Test the PushBack method with another AudioMultiVector as input argument.
+TEST_P(AudioMultiVectorTest, OverwriteAt) {
+  AudioMultiVector vec1(num_channels_);
+  vec1.PushBackInterleaved(array_interleaved_);
+  AudioMultiVector vec2(num_channels_);
+  vec2.Zeros(3);  // 3 zeros in each channel.
+  // Overwrite vec2 at position 5.
+  vec1.OverwriteAt(vec2, 3, 5);
+  // Verify result.
+  // Length remains the same.
+  ASSERT_EQ(array_length(), vec1.Size());
+  int16_t* ptr = array_interleaved_.data();
+  for (size_t i = 0; i < array_length() - 1; ++i) {
+    for (size_t channel = 0; channel < num_channels_; ++channel) {
+      if (i >= 5 && i <= 7) {
+        // Elements 5, 6, 7 should have been replaced with zeros.
+        EXPECT_EQ(0, vec1[channel][i]);
+      } else {
+        EXPECT_EQ(*ptr, vec1[channel][i]);
+      }
+      ++ptr;
+    }
+  }
+}
+
+// Test the CopyChannel method, when the test is instantiated with at least two
+// channels.
+TEST_P(AudioMultiVectorTest, CopyChannel) {
+  if (num_channels_ < 2)
+    return;
+
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved(array_interleaved_);
+  // Create a reference copy.
+  AudioMultiVector ref(num_channels_);
+  ref.PushBack(vec);
+  // Copy from first to last channel.
+  vec.CopyChannel(0, num_channels_ - 1);
+  // Verify that the first and last channels are identical; the others should
+  // be left untouched.
+  for (size_t i = 0; i < array_length(); ++i) {
+    // Verify that all but the last channel are untouched.
+    for (size_t channel = 0; channel < num_channels_ - 1; ++channel) {
+      EXPECT_EQ(ref[channel][i], vec[channel][i]);
+    }
+    // Verify that the last and the first channels are identical.
+    EXPECT_EQ(vec[0][i], vec[num_channels_ - 1][i]);
+  }
+}
+
+TEST_P(AudioMultiVectorTest, PushBackEmptyArray) {
+  AudioMultiVector vec(num_channels_);
+  vec.PushBackInterleaved({});
+  EXPECT_TRUE(vec.Empty());
+}
+
+INSTANTIATE_TEST_SUITE_P(TestNumChannels,
+                         AudioMultiVectorTest,
+                         ::testing::Values(static_cast<size_t>(1),
+                                           static_cast<size_t>(2),
+                                           static_cast<size_t>(5)));
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.cc b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.cc
new file mode 100644
index 0000000000..10e8936447
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.cc
@@ -0,0 +1,381 @@
+/*
+ *  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/neteq/audio_vector.h"
+
+
+#include <algorithm>
+#include <memory>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioVector::AudioVector() : AudioVector(kDefaultInitialSize) {
+  Clear();
+}
+
+AudioVector::AudioVector(size_t initial_size)
+    : array_(new int16_t[initial_size + 1]),
+      capacity_(initial_size + 1),
+      begin_index_(0),
+      end_index_(capacity_ - 1) {
+  memset(array_.get(), 0, capacity_ * sizeof(int16_t));
+}
+
+AudioVector::~AudioVector() = default;
+
+void AudioVector::Clear() {
+  end_index_ = begin_index_ = 0;
+}
+
+void AudioVector::CopyTo(AudioVector* copy_to) const {
+  RTC_DCHECK(copy_to);
+  copy_to->Reserve(Size());
+  CopyTo(Size(), 0, copy_to->array_.get());
+  copy_to->begin_index_ = 0;
+  copy_to->end_index_ = Size();
+}
+
+void AudioVector::CopyTo(size_t length,
+                         size_t position,
+                         int16_t* copy_to) const {
+  if (length == 0)
+    return;
+  length = std::min(length, Size() - position);
+  const size_t copy_index = (begin_index_ + position) % capacity_;
+  const size_t first_chunk_length = std::min(length, capacity_ - copy_index);
+  memcpy(copy_to, &array_[copy_index], first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(&copy_to[first_chunk_length], array_.get(),
+           remaining_length * sizeof(int16_t));
+  }
+}
+
+void AudioVector::PushFront(const AudioVector& prepend_this) {
+  const size_t length = prepend_this.Size();
+  if (length == 0)
+    return;
+
+  // Although the subsequent calling to PushFront does Reserve in it, it is
+  // always more efficient to do a big Reserve first.
+  Reserve(Size() + length);
+
+  const size_t first_chunk_length =
+      std::min(length, prepend_this.capacity_ - prepend_this.begin_index_);
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0)
+    PushFront(prepend_this.array_.get(), remaining_length);
+  PushFront(&prepend_this.array_[prepend_this.begin_index_],
+            first_chunk_length);
+}
+
+void AudioVector::PushFront(const int16_t* prepend_this, size_t length) {
+  if (length == 0)
+    return;
+  Reserve(Size() + length);
+  const size_t first_chunk_length = std::min(length, begin_index_);
+  memcpy(&array_[begin_index_ - first_chunk_length],
+         &prepend_this[length - first_chunk_length],
+         first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(&array_[capacity_ - remaining_length], prepend_this,
+           remaining_length * sizeof(int16_t));
+  }
+  begin_index_ = (begin_index_ + capacity_ - length) % capacity_;
+}
+
+void AudioVector::PushBack(const AudioVector& append_this) {
+  PushBack(append_this, append_this.Size(), 0);
+}
+
+void AudioVector::PushBack(const AudioVector& append_this,
+                           size_t length,
+                           size_t position) {
+  RTC_DCHECK_LE(position, append_this.Size());
+  RTC_DCHECK_LE(length, append_this.Size() - position);
+
+  if (length == 0)
+    return;
+
+  // Although the subsequent calling to PushBack does Reserve in it, it is
+  // always more efficient to do a big Reserve first.
+  Reserve(Size() + length);
+
+  const size_t start_index =
+      (append_this.begin_index_ + position) % append_this.capacity_;
+  const size_t first_chunk_length =
+      std::min(length, append_this.capacity_ - start_index);
+  PushBack(&append_this.array_[start_index], first_chunk_length);
+
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0)
+    PushBack(append_this.array_.get(), remaining_length);
+}
+
+void AudioVector::PushBack(const int16_t* append_this, size_t length) {
+  if (length == 0)
+    return;
+  Reserve(Size() + length);
+  const size_t first_chunk_length = std::min(length, capacity_ - end_index_);
+  memcpy(&array_[end_index_], append_this,
+         first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(array_.get(), &append_this[first_chunk_length],
+           remaining_length * sizeof(int16_t));
+  }
+  end_index_ = (end_index_ + length) % capacity_;
+}
+
+void AudioVector::PopFront(size_t length) {
+  if (length == 0)
+    return;
+  length = std::min(length, Size());
+  begin_index_ = (begin_index_ + length) % capacity_;
+}
+
+void AudioVector::PopBack(size_t length) {
+  if (length == 0)
+    return;
+  // Never remove more than what is in the array.
+  length = std::min(length, Size());
+  end_index_ = (end_index_ + capacity_ - length) % capacity_;
+}
+
+void AudioVector::Extend(size_t extra_length) {
+  if (extra_length == 0)
+    return;
+  InsertZerosByPushBack(extra_length, Size());
+}
+
+void AudioVector::InsertAt(const int16_t* insert_this,
+                           size_t length,
+                           size_t position) {
+  if (length == 0)
+    return;
+  // Cap the insert position at the current array length.
+  position = std::min(Size(), position);
+
+  // When inserting to a position closer to the beginning, it is more efficient
+  // to insert by pushing front than to insert by pushing back, since less data
+  // will be moved, vice versa.
+  if (position <= Size() - position) {
+    InsertByPushFront(insert_this, length, position);
+  } else {
+    InsertByPushBack(insert_this, length, position);
+  }
+}
+
+void AudioVector::InsertZerosAt(size_t length, size_t position) {
+  if (length == 0)
+    return;
+  // Cap the insert position at the current array length.
+  position = std::min(Size(), position);
+
+  // When inserting to a position closer to the beginning, it is more efficient
+  // to insert by pushing front than to insert by pushing back, since less data
+  // will be moved, vice versa.
+  if (position <= Size() - position) {
+    InsertZerosByPushFront(length, position);
+  } else {
+    InsertZerosByPushBack(length, position);
+  }
+}
+
+void AudioVector::OverwriteAt(const AudioVector& insert_this,
+                              size_t length,
+                              size_t position) {
+  RTC_DCHECK_LE(length, insert_this.Size());
+  if (length == 0)
+    return;
+
+  // Cap the insert position at the current array length.
+  position = std::min(Size(), position);
+
+  // Although the subsequent calling to OverwriteAt does Reserve in it, it is
+  // always more efficient to do a big Reserve first.
+  size_t new_size = std::max(Size(), position + length);
+  Reserve(new_size);
+
+  const size_t first_chunk_length =
+      std::min(length, insert_this.capacity_ - insert_this.begin_index_);
+  OverwriteAt(&insert_this.array_[insert_this.begin_index_], first_chunk_length,
+              position);
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    OverwriteAt(insert_this.array_.get(), remaining_length,
+                position + first_chunk_length);
+  }
+}
+
+void AudioVector::OverwriteAt(const int16_t* insert_this,
+                              size_t length,
+                              size_t position) {
+  if (length == 0)
+    return;
+  // Cap the insert position at the current array length.
+  position = std::min(Size(), position);
+
+  size_t new_size = std::max(Size(), position + length);
+  Reserve(new_size);
+
+  const size_t overwrite_index = (begin_index_ + position) % capacity_;
+  const size_t first_chunk_length =
+      std::min(length, capacity_ - overwrite_index);
+  memcpy(&array_[overwrite_index], insert_this,
+         first_chunk_length * sizeof(int16_t));
+  const size_t remaining_length = length - first_chunk_length;
+  if (remaining_length > 0) {
+    memcpy(array_.get(), &insert_this[first_chunk_length],
+           remaining_length * sizeof(int16_t));
+  }
+
+  end_index_ = (begin_index_ + new_size) % capacity_;
+}
+
+void AudioVector::CrossFade(const AudioVector& append_this,
+                            size_t fade_length) {
+  // Fade length cannot be longer than the current vector or `append_this`.
+  RTC_DCHECK_LE(fade_length, Size());
+  RTC_DCHECK_LE(fade_length, append_this.Size());
+  fade_length = std::min(fade_length, Size());
+  fade_length = std::min(fade_length, append_this.Size());
+  size_t position = Size() - fade_length + begin_index_;
+  // Cross fade the overlapping regions.
+  // `alpha` is the mixing factor in Q14.
+  // TODO(hlundin): Consider skipping +1 in the denominator to produce a
+  // smoother cross-fade, in particular at the end of the fade.
+  int alpha_step = 16384 / (static_cast<int>(fade_length) + 1);
+  int alpha = 16384;
+  for (size_t i = 0; i < fade_length; ++i) {
+    alpha -= alpha_step;
+    array_[(position + i) % capacity_] =
+        (alpha * array_[(position + i) % capacity_] +
+         (16384 - alpha) * append_this[i] + 8192) >>
+        14;
+  }
+  RTC_DCHECK_GE(alpha, 0);  // Verify that the slope was correct.
+  // Append what is left of `append_this`.
+  size_t samples_to_push_back = append_this.Size() - fade_length;
+  if (samples_to_push_back > 0)
+    PushBack(append_this, samples_to_push_back, fade_length);
+}
+
+// Returns the number of elements in this AudioVector.
+size_t AudioVector::Size() const {
+  return (end_index_ + capacity_ - begin_index_) % capacity_;
+}
+
+// Returns true if this AudioVector is empty.
+bool AudioVector::Empty() const {
+  return begin_index_ == end_index_;
+}
+
+void AudioVector::Reserve(size_t n) {
+  if (capacity_ > n)
+    return;
+  const size_t length = Size();
+  // Reserve one more sample to remove the ambiguity between empty vector and
+  // full vector. Therefore `begin_index_` == `end_index_` indicates empty
+  // vector, and `begin_index_` == (`end_index_` + 1) % capacity indicates
+  // full vector.
+  std::unique_ptr<int16_t[]> temp_array(new int16_t[n + 1]);
+  CopyTo(length, 0, temp_array.get());
+  array_.swap(temp_array);
+  begin_index_ = 0;
+  end_index_ = length;
+  capacity_ = n + 1;
+}
+
+void AudioVector::InsertByPushBack(const int16_t* insert_this,
+                                   size_t length,
+                                   size_t position) {
+  const size_t move_chunk_length = Size() - position;
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (move_chunk_length > 0) {
+    // TODO(minyue): see if it is possible to avoid copying to a buffer.
+    temp_array.reset(new int16_t[move_chunk_length]);
+    CopyTo(move_chunk_length, position, temp_array.get());
+    PopBack(move_chunk_length);
+  }
+
+  Reserve(Size() + length + move_chunk_length);
+  PushBack(insert_this, length);
+  if (move_chunk_length > 0)
+    PushBack(temp_array.get(), move_chunk_length);
+}
+
+void AudioVector::InsertByPushFront(const int16_t* insert_this,
+                                    size_t length,
+                                    size_t position) {
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (position > 0) {
+    // TODO(minyue): see if it is possible to avoid copying to a buffer.
+    temp_array.reset(new int16_t[position]);
+    CopyTo(position, 0, temp_array.get());
+    PopFront(position);
+  }
+
+  Reserve(Size() + length + position);
+  PushFront(insert_this, length);
+  if (position > 0)
+    PushFront(temp_array.get(), position);
+}
+
+void AudioVector::InsertZerosByPushBack(size_t length, size_t position) {
+  const size_t move_chunk_length = Size() - position;
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (move_chunk_length > 0) {
+    temp_array.reset(new int16_t[move_chunk_length]);
+    CopyTo(move_chunk_length, position, temp_array.get());
+    PopBack(move_chunk_length);
+  }
+
+  Reserve(Size() + length + move_chunk_length);
+
+  const size_t first_zero_chunk_length =
+      std::min(length, capacity_ - end_index_);
+  memset(&array_[end_index_], 0, first_zero_chunk_length * sizeof(int16_t));
+  const size_t remaining_zero_length = length - first_zero_chunk_length;
+  if (remaining_zero_length > 0)
+    memset(array_.get(), 0, remaining_zero_length * sizeof(int16_t));
+  end_index_ = (end_index_ + length) % capacity_;
+
+  if (move_chunk_length > 0)
+    PushBack(temp_array.get(), move_chunk_length);
+}
+
+void AudioVector::InsertZerosByPushFront(size_t length, size_t position) {
+  std::unique_ptr<int16_t[]> temp_array(nullptr);
+  if (position > 0) {
+    temp_array.reset(new int16_t[position]);
+    CopyTo(position, 0, temp_array.get());
+    PopFront(position);
+  }
+
+  Reserve(Size() + length + position);
+
+  const size_t first_zero_chunk_length = std::min(length, begin_index_);
+  memset(&array_[begin_index_ - first_zero_chunk_length], 0,
+         first_zero_chunk_length * sizeof(int16_t));
+  const size_t remaining_zero_length = length - first_zero_chunk_length;
+  if (remaining_zero_length > 0)
+    memset(&array_[capacity_ - remaining_zero_length], 0,
+           remaining_zero_length * sizeof(int16_t));
+  begin_index_ = (begin_index_ + capacity_ - length) % capacity_;
+
+  if (position > 0)
+    PushFront(temp_array.get(), position);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.h b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.h
new file mode 100644
index 0000000000..d68f3ec6be
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector.h
@@ -0,0 +1,172 @@
+/*
+ *  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_NETEQ_AUDIO_VECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
+
+#include <string.h>
+
+#include <cstdint>
+#include <memory>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class AudioVector {
+ public:
+  // Creates an empty AudioVector.
+  AudioVector();
+
+  // Creates an AudioVector with an initial size.
+  explicit AudioVector(size_t initial_size);
+
+  virtual ~AudioVector();
+
+  AudioVector(const AudioVector&) = delete;
+  AudioVector& operator=(const AudioVector&) = delete;
+
+  // Deletes all values and make the vector empty.
+  virtual void Clear();
+
+  // Copies all values from this vector to `copy_to`. Any contents in `copy_to`
+  // are deleted before the copy operation. After the operation is done,
+  // `copy_to` will be an exact replica of this object.
+  virtual void CopyTo(AudioVector* copy_to) const;
+
+  // Copies `length` values from `position` in this vector to `copy_to`.
+  virtual void CopyTo(size_t length, size_t position, int16_t* copy_to) const;
+
+  // Prepends the contents of AudioVector `prepend_this` to this object. The
+  // length of this object is increased with the length of `prepend_this`.
+  virtual void PushFront(const AudioVector& prepend_this);
+
+  // Same as above, but with an array `prepend_this` with `length` elements as
+  // source.
+  virtual void PushFront(const int16_t* prepend_this, size_t length);
+
+  // Same as PushFront but will append to the end of this object.
+  virtual void PushBack(const AudioVector& append_this);
+
+  // Appends a segment of `append_this` to the end of this object. The segment
+  // starts from `position` and has `length` samples.
+  virtual void PushBack(const AudioVector& append_this,
+                        size_t length,
+                        size_t position);
+
+  // Same as PushFront but will append to the end of this object.
+  virtual void PushBack(const int16_t* append_this, size_t length);
+
+  // Removes `length` elements from the beginning of this object.
+  virtual void PopFront(size_t length);
+
+  // Removes `length` elements from the end of this object.
+  virtual void PopBack(size_t length);
+
+  // Extends this object with `extra_length` elements at the end. The new
+  // elements are initialized to zero.
+  virtual void Extend(size_t extra_length);
+
+  // Inserts `length` elements taken from the array `insert_this` and insert
+  // them at `position`. The length of the AudioVector is increased by `length`.
+  // `position` = 0 means that the new values are prepended to the vector.
+  // `position` = Size() means that the new values are appended to the vector.
+  virtual void InsertAt(const int16_t* insert_this,
+                        size_t length,
+                        size_t position);
+
+  // Like InsertAt, but inserts `length` zero elements at `position`.
+  virtual void InsertZerosAt(size_t length, size_t position);
+
+  // Overwrites `length` elements of this AudioVector starting from `position`
+  // with first values in `AudioVector`. The definition of `position`
+  // is the same as for InsertAt(). If `length` and `position` are selected
+  // such that the new data extends beyond the end of the current AudioVector,
+  // the vector is extended to accommodate the new data.
+  virtual void OverwriteAt(const AudioVector& insert_this,
+                           size_t length,
+                           size_t position);
+
+  // Overwrites `length` elements of this AudioVector with values taken from the
+  // array `insert_this`, starting at `position`. The definition of `position`
+  // is the same as for InsertAt(). If `length` and `position` are selected
+  // such that the new data extends beyond the end of the current AudioVector,
+  // the vector is extended to accommodate the new data.
+  virtual void OverwriteAt(const int16_t* insert_this,
+                           size_t length,
+                           size_t position);
+
+  // Appends `append_this` to the end of the current vector. Lets the two
+  // vectors overlap by `fade_length` samples, and cross-fade linearly in this
+  // region.
+  virtual void CrossFade(const AudioVector& append_this, size_t fade_length);
+
+  // Returns the number of elements in this AudioVector.
+  virtual size_t Size() const;
+
+  // Returns true if this AudioVector is empty.
+  virtual bool Empty() const;
+
+  // Accesses and modifies an element of AudioVector.
+  inline const int16_t& operator[](size_t index) const {
+    return array_[WrapIndex(index, begin_index_, capacity_)];
+  }
+
+  inline int16_t& operator[](size_t index) {
+    return array_[WrapIndex(index, begin_index_, capacity_)];
+  }
+
+ private:
+  static const size_t kDefaultInitialSize = 10;
+
+  // This method is used by the [] operators to calculate an index within the
+  // capacity of the array, but without using the modulo operation (%).
+  static inline size_t WrapIndex(size_t index,
+                                 size_t begin_index,
+                                 size_t capacity) {
+    RTC_DCHECK_LT(index, capacity);
+    RTC_DCHECK_LT(begin_index, capacity);
+    size_t ix = begin_index + index;
+    RTC_DCHECK_GE(ix, index);  // Check for overflow.
+    if (ix >= capacity) {
+      ix -= capacity;
+    }
+    RTC_DCHECK_LT(ix, capacity);
+    return ix;
+  }
+
+  void Reserve(size_t n);
+
+  void InsertByPushBack(const int16_t* insert_this,
+                        size_t length,
+                        size_t position);
+
+  void InsertByPushFront(const int16_t* insert_this,
+                         size_t length,
+                         size_t position);
+
+  void InsertZerosByPushBack(size_t length, size_t position);
+
+  void InsertZerosByPushFront(size_t length, size_t position);
+
+  std::unique_ptr<int16_t[]> array_;
+
+  size_t capacity_;  // Allocated number of samples in the array.
+
+  // The index of the first sample in `array_`, except when
+  // |begin_index_ == end_index_|, which indicates an empty buffer.
+  size_t begin_index_;
+
+  // The index of the sample after the last sample in `array_`.
+  size_t end_index_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
new file mode 100644
index 0000000000..ae9dd88606
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
@@ -0,0 +1,384 @@
+/*
+ *  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/neteq/audio_vector.h"
+
+#include <stdlib.h>
+
+#include <string>
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class AudioVectorTest : public ::testing::Test {
+ protected:
+  virtual void SetUp() {
+    // Populate test array.
+    for (size_t i = 0; i < array_length(); ++i) {
+      array_[i] = rtc::checked_cast<int16_t>(i);
+    }
+  }
+
+  size_t array_length() const { return sizeof(array_) / sizeof(array_[0]); }
+
+  int16_t array_[10];
+};
+
+// Create and destroy AudioVector objects, both empty and with a predefined
+// length.
+TEST_F(AudioVectorTest, CreateAndDestroy) {
+  AudioVector vec1;
+  EXPECT_TRUE(vec1.Empty());
+  EXPECT_EQ(0u, vec1.Size());
+
+  size_t initial_size = 17;
+  AudioVector vec2(initial_size);
+  EXPECT_FALSE(vec2.Empty());
+  EXPECT_EQ(initial_size, vec2.Size());
+}
+
+// Test the subscript operator [] for getting and setting.
+TEST_F(AudioVectorTest, SubscriptOperator) {
+  AudioVector vec(array_length());
+  for (size_t i = 0; i < array_length(); ++i) {
+    vec[i] = static_cast<int16_t>(i);
+    const int16_t& value = vec[i];  // Make sure to use the const version.
+    EXPECT_EQ(static_cast<int16_t>(i), value);
+  }
+}
+
+// Test the PushBack method and the CopyFrom method. The Clear method is also
+// invoked.
+TEST_F(AudioVectorTest, PushBackAndCopy) {
+  AudioVector vec;
+  AudioVector vec_copy;
+  vec.PushBack(array_, array_length());
+  vec.CopyTo(&vec_copy);  // Copy from `vec` to `vec_copy`.
+  ASSERT_EQ(array_length(), vec.Size());
+  ASSERT_EQ(array_length(), vec_copy.Size());
+  for (size_t i = 0; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[i]);
+    EXPECT_EQ(array_[i], vec_copy[i]);
+  }
+
+  // Clear `vec` and verify that it is empty.
+  vec.Clear();
+  EXPECT_TRUE(vec.Empty());
+
+  // Now copy the empty vector and verify that the copy becomes empty too.
+  vec.CopyTo(&vec_copy);
+  EXPECT_TRUE(vec_copy.Empty());
+}
+
+// Test the PushBack method with another AudioVector as input argument.
+TEST_F(AudioVectorTest, PushBackVector) {
+  static const size_t kLength = 10;
+  AudioVector vec1(kLength);
+  AudioVector vec2(kLength);
+  // Set the first vector to [0, 1, ..., kLength - 1].
+  // Set the second vector to [kLength, kLength + 1, ..., 2 * kLength - 1].
+  for (size_t i = 0; i < kLength; ++i) {
+    vec1[i] = static_cast<int16_t>(i);
+    vec2[i] = static_cast<int16_t>(i + kLength);
+  }
+  // Append vec2 to the back of vec1.
+  vec1.PushBack(vec2);
+  ASSERT_EQ(2 * kLength, vec1.Size());
+  for (size_t i = 0; i < 2 * kLength; ++i) {
+    EXPECT_EQ(static_cast<int16_t>(i), vec1[i]);
+  }
+}
+
+// Test the PushFront method.
+TEST_F(AudioVectorTest, PushFront) {
+  AudioVector vec;
+  vec.PushFront(array_, array_length());
+  ASSERT_EQ(array_length(), vec.Size());
+  for (size_t i = 0; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[i]);
+  }
+}
+
+// Test the PushFront method with another AudioVector as input argument.
+TEST_F(AudioVectorTest, PushFrontVector) {
+  static const size_t kLength = 10;
+  AudioVector vec1(kLength);
+  AudioVector vec2(kLength);
+  // Set the first vector to [0, 1, ..., kLength - 1].
+  // Set the second vector to [kLength, kLength + 1, ..., 2 * kLength - 1].
+  for (size_t i = 0; i < kLength; ++i) {
+    vec1[i] = static_cast<int16_t>(i);
+    vec2[i] = static_cast<int16_t>(i + kLength);
+  }
+  // Prepend vec1 to the front of vec2.
+  vec2.PushFront(vec1);
+  ASSERT_EQ(2 * kLength, vec2.Size());
+  for (size_t i = 0; i < 2 * kLength; ++i) {
+    EXPECT_EQ(static_cast<int16_t>(i), vec2[i]);
+  }
+}
+
+// Test the PopFront method.
+TEST_F(AudioVectorTest, PopFront) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  vec.PopFront(1);  // Remove one element.
+  EXPECT_EQ(array_length() - 1u, vec.Size());
+  for (size_t i = 0; i < array_length() - 1; ++i) {
+    EXPECT_EQ(static_cast<int16_t>(i + 1), vec[i]);
+  }
+  vec.PopFront(array_length());  // Remove more elements than vector size.
+  EXPECT_EQ(0u, vec.Size());
+}
+
+// Test the PopBack method.
+TEST_F(AudioVectorTest, PopBack) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  vec.PopBack(1);  // Remove one element.
+  EXPECT_EQ(array_length() - 1u, vec.Size());
+  for (size_t i = 0; i < array_length() - 1; ++i) {
+    EXPECT_EQ(static_cast<int16_t>(i), vec[i]);
+  }
+  vec.PopBack(array_length());  // Remove more elements than vector size.
+  EXPECT_EQ(0u, vec.Size());
+}
+
+// Test the Extend method.
+TEST_F(AudioVectorTest, Extend) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  vec.Extend(5);  // Extend with 5 elements, which should all be zeros.
+  ASSERT_EQ(array_length() + 5u, vec.Size());
+  // Verify that all are zero.
+  for (size_t i = array_length(); i < array_length() + 5; ++i) {
+    EXPECT_EQ(0, vec[i]);
+  }
+}
+
+// Test the InsertAt method with an insert position in the middle of the vector.
+TEST_F(AudioVectorTest, InsertAt) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  int insert_position = 5;
+  vec.InsertAt(new_array, kNewLength, insert_position);
+  // Verify that the vector looks as follows:
+  // {0, 1, ..., `insert_position` - 1, 100, 101, ..., 100 + kNewLength - 1,
+  //  `insert_position`, `insert_position` + 1, ..., kLength - 1}.
+  size_t pos = 0;
+  for (int i = 0; i < insert_position; ++i) {
+    EXPECT_EQ(array_[i], vec[pos]);
+    ++pos;
+  }
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+  for (size_t i = insert_position; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[pos]);
+    ++pos;
+  }
+}
+
+// Test the InsertZerosAt method with an insert position in the middle of the
+// vector. Use the InsertAt method as reference.
+TEST_F(AudioVectorTest, InsertZerosAt) {
+  AudioVector vec;
+  AudioVector vec_ref;
+  vec.PushBack(array_, array_length());
+  vec_ref.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int insert_position = 5;
+  vec.InsertZerosAt(kNewLength, insert_position);
+  int16_t new_array[kNewLength] = {0};  // All zero elements.
+  vec_ref.InsertAt(new_array, kNewLength, insert_position);
+  // Verify that the vectors are identical.
+  ASSERT_EQ(vec_ref.Size(), vec.Size());
+  for (size_t i = 0; i < vec.Size(); ++i) {
+    EXPECT_EQ(vec_ref[i], vec[i]);
+  }
+}
+
+// Test the InsertAt method with an insert position at the start of the vector.
+TEST_F(AudioVectorTest, InsertAtBeginning) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  int insert_position = 0;
+  vec.InsertAt(new_array, kNewLength, insert_position);
+  // Verify that the vector looks as follows:
+  // {100, 101, ..., 100 + kNewLength - 1,
+  //  0, 1, ..., kLength - 1}.
+  size_t pos = 0;
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+  for (size_t i = insert_position; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[pos]);
+    ++pos;
+  }
+}
+
+// Test the InsertAt method with an insert position at the end of the vector.
+TEST_F(AudioVectorTest, InsertAtEnd) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  int insert_position = rtc::checked_cast<int>(array_length());
+  vec.InsertAt(new_array, kNewLength, insert_position);
+  // Verify that the vector looks as follows:
+  // {0, 1, ..., kLength - 1, 100, 101, ..., 100 + kNewLength - 1 }.
+  size_t pos = 0;
+  for (size_t i = 0; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[pos]);
+    ++pos;
+  }
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+}
+
+// Test the InsertAt method with an insert position beyond the end of the
+// vector. Verify that a position beyond the end of the vector does not lead to
+// an error. The expected outcome is the same as if the vector end was used as
+// input position. That is, the input position should be capped at the maximum
+// allowed value.
+TEST_F(AudioVectorTest, InsertBeyondEnd) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  int insert_position =
+      rtc::checked_cast<int>(array_length() + 10);  // Too large.
+  vec.InsertAt(new_array, kNewLength, insert_position);
+  // Verify that the vector looks as follows:
+  // {0, 1, ..., kLength - 1, 100, 101, ..., 100 + kNewLength - 1 }.
+  size_t pos = 0;
+  for (size_t i = 0; i < array_length(); ++i) {
+    EXPECT_EQ(array_[i], vec[pos]);
+    ++pos;
+  }
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+}
+
+// Test the OverwriteAt method with a position such that all of the new values
+// fit within the old vector.
+TEST_F(AudioVectorTest, OverwriteAt) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  size_t insert_position = 2;
+  vec.OverwriteAt(new_array, kNewLength, insert_position);
+  // Verify that the vector looks as follows:
+  // {0, ..., `insert_position` - 1, 100, 101, ..., 100 + kNewLength - 1,
+  //  `insert_position`, `insert_position` + 1, ..., kLength - 1}.
+  size_t pos = 0;
+  for (pos = 0; pos < insert_position; ++pos) {
+    EXPECT_EQ(array_[pos], vec[pos]);
+  }
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+  for (; pos < array_length(); ++pos) {
+    EXPECT_EQ(array_[pos], vec[pos]);
+  }
+}
+
+// Test the OverwriteAt method with a position such that some of the new values
+// extend beyond the end of the current vector. This is valid, and the vector is
+// expected to expand to accommodate the new values.
+TEST_F(AudioVectorTest, OverwriteBeyondEnd) {
+  AudioVector vec;
+  vec.PushBack(array_, array_length());
+  static const int kNewLength = 5;
+  int16_t new_array[kNewLength];
+  // Set array elements to {100, 101, 102, ... }.
+  for (int i = 0; i < kNewLength; ++i) {
+    new_array[i] = 100 + i;
+  }
+  int insert_position = rtc::checked_cast<int>(array_length() - 2);
+  vec.OverwriteAt(new_array, kNewLength, insert_position);
+  ASSERT_EQ(array_length() - 2u + kNewLength, vec.Size());
+  // Verify that the vector looks as follows:
+  // {0, ..., `insert_position` - 1, 100, 101, ..., 100 + kNewLength - 1,
+  //  `insert_position`, `insert_position` + 1, ..., kLength - 1}.
+  int pos = 0;
+  for (pos = 0; pos < insert_position; ++pos) {
+    EXPECT_EQ(array_[pos], vec[pos]);
+  }
+  for (int i = 0; i < kNewLength; ++i) {
+    EXPECT_EQ(new_array[i], vec[pos]);
+    ++pos;
+  }
+  // Verify that we checked to the end of `vec`.
+  EXPECT_EQ(vec.Size(), static_cast<size_t>(pos));
+}
+
+TEST_F(AudioVectorTest, CrossFade) {
+  static const size_t kLength = 100;
+  static const size_t kFadeLength = 10;
+  AudioVector vec1(kLength);
+  AudioVector vec2(kLength);
+  // Set all vector elements to 0 in `vec1` and 100 in `vec2`.
+  for (size_t i = 0; i < kLength; ++i) {
+    vec1[i] = 0;
+    vec2[i] = 100;
+  }
+  vec1.CrossFade(vec2, kFadeLength);
+  ASSERT_EQ(2 * kLength - kFadeLength, vec1.Size());
+  // First part untouched.
+  for (size_t i = 0; i < kLength - kFadeLength; ++i) {
+    EXPECT_EQ(0, vec1[i]);
+  }
+  // Check mixing zone.
+  for (size_t i = 0; i < kFadeLength; ++i) {
+    EXPECT_NEAR((i + 1) * 100 / (kFadeLength + 1),
+                vec1[kLength - kFadeLength + i], 1);
+  }
+  // Second part untouched.
+  for (size_t i = kLength; i < vec1.Size(); ++i) {
+    EXPECT_EQ(100, vec1[i]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc
new file mode 100644
index 0000000000..2c95d3b390
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc
@@ -0,0 +1,309 @@
+/*
+ *  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/neteq/background_noise.h"
+
+#include <string.h>  // memcpy
+
+#include <algorithm>  // min, max
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/post_decode_vad.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kMaxSampleRate = 48000;
+
+}  // namespace
+
+// static
+constexpr size_t BackgroundNoise::kMaxLpcOrder;
+
+BackgroundNoise::BackgroundNoise(size_t num_channels)
+    : num_channels_(num_channels),
+      channel_parameters_(new ChannelParameters[num_channels_]) {
+  Reset();
+}
+
+BackgroundNoise::~BackgroundNoise() {}
+
+void BackgroundNoise::Reset() {
+  initialized_ = false;
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    channel_parameters_[channel].Reset();
+  }
+}
+
+bool BackgroundNoise::Update(const AudioMultiVector& input,
+                             const PostDecodeVad& vad) {
+  bool filter_params_saved = false;
+  if (vad.running() && vad.active_speech()) {
+    // Do not update the background noise parameters if we know that the signal
+    // is active speech.
+    return filter_params_saved;
+  }
+
+  int32_t auto_correlation[kMaxLpcOrder + 1];
+  int16_t fiter_output[kMaxLpcOrder + kResidualLength];
+  int16_t reflection_coefficients[kMaxLpcOrder];
+  int16_t lpc_coefficients[kMaxLpcOrder + 1];
+
+  for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
+    ChannelParameters& parameters = channel_parameters_[channel_ix];
+    int16_t temp_signal_array[kVecLen + kMaxLpcOrder] = {0};
+    int16_t* temp_signal = &temp_signal_array[kMaxLpcOrder];
+    RTC_DCHECK_GE(input.Size(), kVecLen);
+    input[channel_ix].CopyTo(kVecLen, input.Size() - kVecLen, temp_signal);
+    int32_t sample_energy =
+        CalculateAutoCorrelation(temp_signal, kVecLen, auto_correlation);
+
+    if ((!vad.running() &&
+         sample_energy < parameters.energy_update_threshold) ||
+        (vad.running() && !vad.active_speech())) {
+      // Generate LPC coefficients.
+      if (auto_correlation[0] <= 0) {
+        // Center value in auto-correlation is not positive. Do not update.
+        return filter_params_saved;
+      }
+
+      // Regardless of whether the filter is actually updated or not,
+      // update energy threshold levels, since we have in fact observed
+      // a low energy signal.
+      if (sample_energy < parameters.energy_update_threshold) {
+        // Never go under 1.0 in average sample energy.
+        parameters.energy_update_threshold = std::max(sample_energy, 1);
+        parameters.low_energy_update_threshold = 0;
+      }
+
+      // Only update BGN if filter is stable, i.e., if return value from
+      // Levinson-Durbin function is 1.
+      if (WebRtcSpl_LevinsonDurbin(auto_correlation, lpc_coefficients,
+                                   reflection_coefficients,
+                                   kMaxLpcOrder) != 1) {
+        return filter_params_saved;
+      }
+
+      // Generate the CNG gain factor by looking at the energy of the residual.
+      WebRtcSpl_FilterMAFastQ12(temp_signal + kVecLen - kResidualLength,
+                                fiter_output, lpc_coefficients,
+                                kMaxLpcOrder + 1, kResidualLength);
+      int32_t residual_energy = WebRtcSpl_DotProductWithScale(
+          fiter_output, fiter_output, kResidualLength, 0);
+
+      // Check spectral flatness.
+      // Comparing the residual variance with the input signal variance tells
+      // if the spectrum is flat or not.
+      // If 5 * residual_energy >= 16 * sample_energy, the spectrum is flat
+      // enough.  Also ensure that the energy is non-zero.
+      if ((sample_energy > 0) &&
+          (int64_t{5} * residual_energy >= int64_t{16} * sample_energy)) {
+        // Spectrum is flat enough; save filter parameters.
+        // `temp_signal` + `kVecLen` - `kMaxLpcOrder` points at the first of the
+        // `kMaxLpcOrder` samples in the residual signal, which will form the
+        // filter state for the next noise generation.
+        SaveParameters(channel_ix, lpc_coefficients,
+                       temp_signal + kVecLen - kMaxLpcOrder, sample_energy,
+                       residual_energy);
+        filter_params_saved = true;
+      }
+    } else {
+      // Will only happen if post-decode VAD is disabled and `sample_energy` is
+      // not low enough. Increase the threshold for update so that it increases
+      // by a factor 4 in 4 seconds.
+      IncrementEnergyThreshold(channel_ix, sample_energy);
+    }
+  }
+  return filter_params_saved;
+}
+
+void BackgroundNoise::GenerateBackgroundNoise(
+    rtc::ArrayView<const int16_t> random_vector,
+    size_t channel,
+    int mute_slope,
+    bool too_many_expands,
+    size_t num_noise_samples,
+    int16_t* buffer) {
+  constexpr size_t kNoiseLpcOrder = kMaxLpcOrder;
+  int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125];
+  RTC_DCHECK_LE(num_noise_samples, (kMaxSampleRate / 8000 * 125));
+  RTC_DCHECK_GE(random_vector.size(), num_noise_samples);
+  int16_t* noise_samples = &buffer[kNoiseLpcOrder];
+  if (initialized()) {
+    // Use background noise parameters.
+    memcpy(noise_samples - kNoiseLpcOrder, FilterState(channel),
+           sizeof(int16_t) * kNoiseLpcOrder);
+
+    int dc_offset = 0;
+    if (ScaleShift(channel) > 1) {
+      dc_offset = 1 << (ScaleShift(channel) - 1);
+    }
+
+    // Scale random vector to correct energy level.
+    WebRtcSpl_AffineTransformVector(scaled_random_vector, random_vector.data(),
+                                    Scale(channel), dc_offset,
+                                    ScaleShift(channel), num_noise_samples);
+
+    WebRtcSpl_FilterARFastQ12(scaled_random_vector, noise_samples,
+                              Filter(channel), kNoiseLpcOrder + 1,
+                              num_noise_samples);
+
+    SetFilterState(
+        channel,
+        {&(noise_samples[num_noise_samples - kNoiseLpcOrder]), kNoiseLpcOrder});
+
+    // Unmute the background noise.
+    int16_t bgn_mute_factor = MuteFactor(channel);
+    if (bgn_mute_factor < 16384) {
+      WebRtcSpl_AffineTransformVector(noise_samples, noise_samples,
+                                      bgn_mute_factor, 8192, 14,
+                                      num_noise_samples);
+    }
+    // Update mute_factor in BackgroundNoise class.
+    SetMuteFactor(channel, bgn_mute_factor);
+  } else {
+    // BGN parameters have not been initialized; use zero noise.
+    memset(noise_samples, 0, sizeof(int16_t) * num_noise_samples);
+  }
+}
+
+int32_t BackgroundNoise::Energy(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].energy;
+}
+
+void BackgroundNoise::SetMuteFactor(size_t channel, int16_t value) {
+  RTC_DCHECK_LT(channel, num_channels_);
+  channel_parameters_[channel].mute_factor = value;
+}
+
+int16_t BackgroundNoise::MuteFactor(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].mute_factor;
+}
+
+const int16_t* BackgroundNoise::Filter(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].filter;
+}
+
+const int16_t* BackgroundNoise::FilterState(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].filter_state;
+}
+
+void BackgroundNoise::SetFilterState(size_t channel,
+                                     rtc::ArrayView<const int16_t> input) {
+  RTC_DCHECK_LT(channel, num_channels_);
+  size_t length = std::min(input.size(), kMaxLpcOrder);
+  memcpy(channel_parameters_[channel].filter_state, input.data(),
+         length * sizeof(int16_t));
+}
+
+int16_t BackgroundNoise::Scale(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].scale;
+}
+int16_t BackgroundNoise::ScaleShift(size_t channel) const {
+  RTC_DCHECK_LT(channel, num_channels_);
+  return channel_parameters_[channel].scale_shift;
+}
+
+int32_t BackgroundNoise::CalculateAutoCorrelation(
+    const int16_t* signal,
+    size_t length,
+    int32_t* auto_correlation) const {
+  static const int kCorrelationStep = -1;
+  const int correlation_scale =
+      CrossCorrelationWithAutoShift(signal, signal, length, kMaxLpcOrder + 1,
+                                    kCorrelationStep, auto_correlation);
+
+  // Number of shifts to normalize energy to energy/sample.
+  int energy_sample_shift = kLogVecLen - correlation_scale;
+  return auto_correlation[0] >> energy_sample_shift;
+}
+
+void BackgroundNoise::IncrementEnergyThreshold(size_t channel,
+                                               int32_t sample_energy) {
+  // TODO(hlundin): Simplify the below threshold update. What this code
+  // does is simply "threshold += (increment * threshold) >> 16", but due
+  // to the limited-width operations, it is not exactly the same. The
+  // difference should be inaudible, but bit-exactness would not be
+  // maintained.
+  RTC_DCHECK_LT(channel, num_channels_);
+  ChannelParameters& parameters = channel_parameters_[channel];
+  int32_t temp_energy =
+      (kThresholdIncrement * parameters.low_energy_update_threshold) >> 16;
+  temp_energy +=
+      kThresholdIncrement * (parameters.energy_update_threshold & 0xFF);
+  temp_energy +=
+      (kThresholdIncrement * ((parameters.energy_update_threshold >> 8) & 0xFF))
+      << 8;
+  parameters.low_energy_update_threshold += temp_energy;
+
+  parameters.energy_update_threshold +=
+      kThresholdIncrement * (parameters.energy_update_threshold >> 16);
+  parameters.energy_update_threshold +=
+      parameters.low_energy_update_threshold >> 16;
+  parameters.low_energy_update_threshold =
+      parameters.low_energy_update_threshold & 0x0FFFF;
+
+  // Update maximum energy.
+  // Decrease by a factor 1/1024 each time.
+  parameters.max_energy = parameters.max_energy - (parameters.max_energy >> 10);
+  if (sample_energy > parameters.max_energy) {
+    parameters.max_energy = sample_energy;
+  }
+
+  // Set `energy_update_threshold` to no less than 60 dB lower than
+  // `max_energy_`. Adding 524288 assures proper rounding.
+  int32_t energy_update_threshold = (parameters.max_energy + 524288) >> 20;
+  if (energy_update_threshold > parameters.energy_update_threshold) {
+    parameters.energy_update_threshold = energy_update_threshold;
+  }
+}
+
+void BackgroundNoise::SaveParameters(size_t channel,
+                                     const int16_t* lpc_coefficients,
+                                     const int16_t* filter_state,
+                                     int32_t sample_energy,
+                                     int32_t residual_energy) {
+  RTC_DCHECK_LT(channel, num_channels_);
+  ChannelParameters& parameters = channel_parameters_[channel];
+  memcpy(parameters.filter, lpc_coefficients,
+         (kMaxLpcOrder + 1) * sizeof(int16_t));
+  memcpy(parameters.filter_state, filter_state, kMaxLpcOrder * sizeof(int16_t));
+  // Save energy level and update energy threshold levels.
+  // Never get under 1.0 in average sample energy.
+  parameters.energy = std::max(sample_energy, 1);
+  parameters.energy_update_threshold = parameters.energy;
+  parameters.low_energy_update_threshold = 0;
+
+  // Normalize residual_energy to 29 or 30 bits before sqrt.
+  int16_t norm_shift = WebRtcSpl_NormW32(residual_energy) - 1;
+  if (norm_shift & 0x1) {
+    norm_shift -= 1;  // Even number of shifts required.
+  }
+  residual_energy = WEBRTC_SPL_SHIFT_W32(residual_energy, norm_shift);
+
+  // Calculate scale and shift factor.
+  parameters.scale = static_cast<int16_t>(WebRtcSpl_SqrtFloor(residual_energy));
+  // Add 13 to the `scale_shift_`, since the random numbers table is in
+  // Q13.
+  // TODO(hlundin): Move the "13" to where the `scale_shift_` is used?
+  parameters.scale_shift =
+      static_cast<int16_t>(13 + ((kLogResidualLength + norm_shift) / 2));
+
+  initialized_ = true;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.h b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.h
new file mode 100644
index 0000000000..8e6d5890a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.h
@@ -0,0 +1,138 @@
+/*
+ *  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_NETEQ_BACKGROUND_NOISE_H_
+#define MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
+
+#include <string.h>  // size_t
+
+#include <memory>
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class AudioMultiVector;
+class PostDecodeVad;
+
+// This class handles estimation of background noise parameters.
+class BackgroundNoise {
+ public:
+  // TODO(hlundin): For 48 kHz support, increase kMaxLpcOrder to 10.
+  // Will work anyway, but probably sound a little worse.
+  static constexpr size_t kMaxLpcOrder = 8;  // 32000 / 8000 + 4.
+
+  explicit BackgroundNoise(size_t num_channels);
+  virtual ~BackgroundNoise();
+
+  BackgroundNoise(const BackgroundNoise&) = delete;
+  BackgroundNoise& operator=(const BackgroundNoise&) = delete;
+
+  void Reset();
+
+  // Updates the parameter estimates based on the signal currently in the
+  // `sync_buffer`, and on the latest decision in `vad` if it is running.
+  // Returns true if the filter parameters are updated.
+  bool Update(const AudioMultiVector& sync_buffer, const PostDecodeVad& vad);
+
+  // Generates background noise given a random vector and writes the output to
+  // `buffer`.
+  void GenerateBackgroundNoise(rtc::ArrayView<const int16_t> random_vector,
+                               size_t channel,
+                               int mute_slope,
+                               bool too_many_expands,
+                               size_t num_noise_samples,
+                               int16_t* buffer);
+
+  // Returns `energy_` for `channel`.
+  int32_t Energy(size_t channel) const;
+
+  // Sets the value of `mute_factor_` for `channel` to `value`.
+  void SetMuteFactor(size_t channel, int16_t value);
+
+  // Returns `mute_factor_` for `channel`.
+  int16_t MuteFactor(size_t channel) const;
+
+  // Returns a pointer to `filter_` for `channel`.
+  const int16_t* Filter(size_t channel) const;
+
+  // Returns a pointer to `filter_state_` for `channel`.
+  const int16_t* FilterState(size_t channel) const;
+
+  // Copies `input` to the filter state. Will not copy more than `kMaxLpcOrder`
+  // elements.
+  void SetFilterState(size_t channel, rtc::ArrayView<const int16_t> input);
+
+  // Returns `scale_` for `channel`.
+  int16_t Scale(size_t channel) const;
+
+  // Returns `scale_shift_` for `channel`.
+  int16_t ScaleShift(size_t channel) const;
+
+  // Accessors.
+  bool initialized() const { return initialized_; }
+
+ private:
+  static const int kThresholdIncrement = 229;  // 0.0035 in Q16.
+  static const size_t kVecLen = 256;
+  static const int kLogVecLen = 8;  // log2(kVecLen).
+  static const size_t kResidualLength = 64;
+  static const int16_t kLogResidualLength = 6;  // log2(kResidualLength)
+
+  struct ChannelParameters {
+    // Constructor.
+    ChannelParameters() { Reset(); }
+
+    void Reset() {
+      energy = 2500;
+      max_energy = 0;
+      energy_update_threshold = 500000;
+      low_energy_update_threshold = 0;
+      memset(filter_state, 0, sizeof(filter_state));
+      memset(filter, 0, sizeof(filter));
+      filter[0] = 4096;
+      mute_factor = 0;
+      scale = 20000;
+      scale_shift = 24;
+    }
+
+    int32_t energy;
+    int32_t max_energy;
+    int32_t energy_update_threshold;
+    int32_t low_energy_update_threshold;
+    int16_t filter_state[kMaxLpcOrder];
+    int16_t filter[kMaxLpcOrder + 1];
+    int16_t mute_factor;
+    int16_t scale;
+    int16_t scale_shift;
+  };
+
+  int32_t CalculateAutoCorrelation(const int16_t* signal,
+                                   size_t length,
+                                   int32_t* auto_correlation) const;
+
+  // Increments the energy threshold by a factor 1 + `kThresholdIncrement`.
+  void IncrementEnergyThreshold(size_t channel, int32_t sample_energy);
+
+  // Updates the filter parameters.
+  void SaveParameters(size_t channel,
+                      const int16_t* lpc_coefficients,
+                      const int16_t* filter_state,
+                      int32_t sample_energy,
+                      int32_t residual_energy);
+
+  size_t num_channels_;
+  std::unique_ptr<ChannelParameters[]> channel_parameters_;
+  bool initialized_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/background_noise_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise_unittest.cc
new file mode 100644
index 0000000000..e32492f57e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise_unittest.cc
@@ -0,0 +1,26 @@
+/*
+ *  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.
+ */
+
+// Unit tests for BackgroundNoise class.
+
+#include "modules/audio_coding/neteq/background_noise.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(BackgroundNoise, CreateAndDestroy) {
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.cc
new file mode 100644
index 0000000000..2c42d0d13f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.cc
@@ -0,0 +1,64 @@
+/*
+ *  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/neteq/buffer_level_filter.h"
+
+#include <stdint.h>
+
+#include <algorithm>
+
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+BufferLevelFilter::BufferLevelFilter() {
+  Reset();
+}
+
+void BufferLevelFilter::Reset() {
+  filtered_current_level_ = 0;
+  level_factor_ = 253;
+}
+
+void BufferLevelFilter::Update(size_t buffer_size_samples,
+                               int time_stretched_samples) {
+  // Filter:
+  // `filtered_current_level_` = `level_factor_` * `filtered_current_level_` +
+  //                            (1 - `level_factor_`) * `buffer_size_samples`
+  // `level_factor_` and `filtered_current_level_` are in Q8.
+  // `buffer_size_samples` is in Q0.
+  const int64_t filtered_current_level =
+      (level_factor_ * int64_t{filtered_current_level_} >> 8) +
+      (256 - level_factor_) * rtc::dchecked_cast<int64_t>(buffer_size_samples);
+
+  // Account for time-scale operations (accelerate and pre-emptive expand) and
+  // make sure that the filtered value remains non-negative.
+  filtered_current_level_ = rtc::saturated_cast<int>(std::max<int64_t>(
+      0, filtered_current_level - int64_t{time_stretched_samples} * (1 << 8)));
+}
+
+void BufferLevelFilter::SetFilteredBufferLevel(int buffer_size_samples) {
+  filtered_current_level_ =
+      rtc::saturated_cast<int>(int64_t{buffer_size_samples} * 256);
+}
+
+void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level_ms) {
+  if (target_buffer_level_ms <= 20) {
+    level_factor_ = 251;
+  } else if (target_buffer_level_ms <= 60) {
+    level_factor_ = 252;
+  } else if (target_buffer_level_ms <= 140) {
+    level_factor_ = 253;
+  } else {
+    level_factor_ = 254;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.h b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.h
new file mode 100644
index 0000000000..ced36da9c2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.h
@@ -0,0 +1,54 @@
+/*
+ *  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_NETEQ_BUFFER_LEVEL_FILTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+class BufferLevelFilter {
+ public:
+  BufferLevelFilter();
+  virtual ~BufferLevelFilter() {}
+
+  BufferLevelFilter(const BufferLevelFilter&) = delete;
+  BufferLevelFilter& operator=(const BufferLevelFilter&) = delete;
+
+  virtual void Reset();
+
+  // Updates the filter. Current buffer size is `buffer_size_samples`.
+  // `time_stretched_samples` is subtracted from the filtered value (thus
+  // bypassing the filter operation).
+  virtual void Update(size_t buffer_size_samples, int time_stretched_samples);
+
+  // Set the filtered buffer level to a particular value directly. This should
+  // only be used in case of large changes in buffer size, such as buffer
+  // flushes.
+  virtual void SetFilteredBufferLevel(int buffer_size_samples);
+
+  // The target level is used to select the appropriate filter coefficient.
+  virtual void SetTargetBufferLevel(int target_buffer_level_ms);
+
+  // Returns filtered current level in number of samples.
+  virtual int filtered_current_level() const {
+    // Round to nearest whole sample.
+    return (int64_t{filtered_current_level_} + (1 << 7)) >> 8;
+  }
+
+ private:
+  int level_factor_;  // Filter factor for the buffer level filter in Q8.
+  int filtered_current_level_;  // Filtered current buffer level in Q8.
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
new file mode 100644
index 0000000000..6773e96f58
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
@@ -0,0 +1,116 @@
+/*
+ *  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.
+ */
+
+// Unit tests for BufferLevelFilter class.
+
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+
+#include <math.h>  // Access to pow function.
+
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(BufferLevelFilter, CreateAndDestroy) {
+  BufferLevelFilter* filter = new BufferLevelFilter();
+  EXPECT_EQ(0, filter->filtered_current_level());
+  delete filter;
+}
+
+TEST(BufferLevelFilter, ConvergenceTest) {
+  BufferLevelFilter filter;
+  for (int times = 10; times <= 50; times += 10) {
+    for (int value = 100; value <= 200; value += 10) {
+      filter.Reset();
+      filter.SetTargetBufferLevel(20);  // Makes filter coefficient 251/256.
+      rtc::StringBuilder ss;
+      ss << "times = " << times << ", value = " << value;
+      SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+      for (int i = 0; i < times; ++i) {
+        filter.Update(value, 0 /* time_stretched_samples */);
+      }
+      // Expect the filtered value to be (theoretically)
+      // (1 - (251/256) ^ `times`) * `value`.
+      double expected_value_double = (1 - pow(251.0 / 256.0, times)) * value;
+      int expected_value = static_cast<int>(expected_value_double);
+
+      // The actual value may differ slightly from the expected value due to
+      // intermediate-stage rounding errors in the filter implementation.
+      // This is why we have to use EXPECT_NEAR with a tolerance of +/-1.
+      EXPECT_NEAR(expected_value, filter.filtered_current_level(), 1);
+    }
+  }
+}
+
+// Verify that target buffer level impacts on the filter convergence.
+TEST(BufferLevelFilter, FilterFactor) {
+  BufferLevelFilter filter;
+  // Update 10 times with value 100.
+  const int kTimes = 10;
+  const int kValue = 100;
+
+  filter.SetTargetBufferLevel(60);  // Makes filter coefficient 252/256.
+  for (int i = 0; i < kTimes; ++i) {
+    filter.Update(kValue, 0 /* time_stretched_samples */);
+  }
+  // Expect the filtered value to be
+  // (1 - (252/256) ^ `kTimes`) * `kValue`.
+  int expected_value = 15;
+  EXPECT_EQ(expected_value, filter.filtered_current_level());
+
+  filter.Reset();
+  filter.SetTargetBufferLevel(140);  // Makes filter coefficient 253/256.
+  for (int i = 0; i < kTimes; ++i) {
+    filter.Update(kValue, 0 /* time_stretched_samples */);
+  }
+  // Expect the filtered value to be
+  // (1 - (253/256) ^ `kTimes`) * `kValue`.
+  expected_value = 11;
+  EXPECT_EQ(expected_value, filter.filtered_current_level());
+
+  filter.Reset();
+  filter.SetTargetBufferLevel(160);  // Makes filter coefficient 254/256.
+  for (int i = 0; i < kTimes; ++i) {
+    filter.Update(kValue, 0 /* time_stretched_samples */);
+  }
+  // Expect the filtered value to be
+  // (1 - (254/256) ^ `kTimes`) * `kValue`.
+  expected_value = 8;
+  EXPECT_EQ(expected_value, filter.filtered_current_level());
+}
+
+TEST(BufferLevelFilter, TimeStretchedSamples) {
+  BufferLevelFilter filter;
+  filter.SetTargetBufferLevel(20);  // Makes filter coefficient 251/256.
+  // Update 10 times with value 100.
+  const int kTimes = 10;
+  const int kValue = 100;
+  const int kTimeStretchedSamples = 3;
+  for (int i = 0; i < kTimes; ++i) {
+    filter.Update(kValue, 0);
+  }
+  // Expect the filtered value to be
+  // (1 - (251/256) ^ `kTimes`) * `kValue`.
+  const int kExpectedValue = 18;
+  EXPECT_EQ(kExpectedValue, filter.filtered_current_level());
+
+  // Update filter again, now with non-zero value for packet length.
+  // Set the current filtered value to be the input, in order to isolate the
+  // impact of `kTimeStretchedSamples`.
+  filter.Update(filter.filtered_current_level(), kTimeStretchedSamples);
+  EXPECT_EQ(kExpectedValue - kTimeStretchedSamples,
+            filter.filtered_current_level());
+  // Try negative value and verify that we come back to the previous result.
+  filter.Update(filter.filtered_current_level(), -kTimeStretchedSamples);
+  EXPECT_EQ(kExpectedValue, filter.filtered_current_level());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.cc b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.cc
new file mode 100644
index 0000000000..a2ce888f45
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.cc
@@ -0,0 +1,130 @@
+/*
+ *  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/neteq/comfort_noise.h"
+
+
+#include <cstdint>
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+void ComfortNoise::Reset() {
+  first_call_ = true;
+}
+
+int ComfortNoise::UpdateParameters(const Packet& packet) {
+  // Get comfort noise decoder.
+  if (decoder_database_->SetActiveCngDecoder(packet.payload_type) != kOK) {
+    return kUnknownPayloadType;
+  }
+  ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+  RTC_DCHECK(cng_decoder);
+  cng_decoder->UpdateSid(packet.payload);
+  return kOK;
+}
+
+int ComfortNoise::Generate(size_t requested_length, AudioMultiVector* output) {
+  // TODO(hlundin): Change to an enumerator and skip assert.
+  RTC_DCHECK(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
+             fs_hz_ == 48000);
+  // Not adapted for multi-channel yet.
+  if (output->Channels() != 1) {
+    RTC_LOG(LS_ERROR) << "No multi-channel support";
+    return kMultiChannelNotSupported;
+  }
+
+  size_t number_of_samples = requested_length;
+  bool new_period = false;
+  if (first_call_) {
+    // Generate noise and overlap slightly with old data.
+    number_of_samples = requested_length + overlap_length_;
+    new_period = true;
+  }
+  output->AssertSize(number_of_samples);
+  // Get the decoder from the database.
+  ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+  if (!cng_decoder) {
+    RTC_LOG(LS_ERROR) << "Unknwown payload type";
+    return kUnknownPayloadType;
+  }
+
+  std::unique_ptr<int16_t[]> temp(new int16_t[number_of_samples]);
+  if (!cng_decoder->Generate(
+          rtc::ArrayView<int16_t>(temp.get(), number_of_samples), new_period)) {
+    // Error returned.
+    output->Zeros(requested_length);
+    RTC_LOG(LS_ERROR)
+        << "ComfortNoiseDecoder::Genererate failed to generate comfort noise";
+    return kInternalError;
+  }
+  (*output)[0].OverwriteAt(temp.get(), number_of_samples, 0);
+
+  if (first_call_) {
+    // Set tapering window parameters. Values are in Q15.
+    int16_t muting_window;              // Mixing factor for overlap data.
+    int16_t muting_window_increment;    // Mixing factor increment (negative).
+    int16_t unmuting_window;            // Mixing factor for comfort noise.
+    int16_t unmuting_window_increment;  // Mixing factor increment.
+    if (fs_hz_ == 8000) {
+      muting_window = DspHelper::kMuteFactorStart8kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement8kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart8kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement8kHz;
+    } else if (fs_hz_ == 16000) {
+      muting_window = DspHelper::kMuteFactorStart16kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement16kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart16kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement16kHz;
+    } else if (fs_hz_ == 32000) {
+      muting_window = DspHelper::kMuteFactorStart32kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement32kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart32kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement32kHz;
+    } else {  // fs_hz_ == 48000
+      muting_window = DspHelper::kMuteFactorStart48kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement48kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart48kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement48kHz;
+    }
+
+    // Do overlap-add between new vector and overlap.
+    size_t start_ix = sync_buffer_->Size() - overlap_length_;
+    for (size_t i = 0; i < overlap_length_; i++) {
+      /* overlapVec[i] = WinMute * overlapVec[i] + WinUnMute * outData[i] */
+      // The expression (*output)[0][i] is the i-th element in the first
+      // channel.
+      (*sync_buffer_)[0][start_ix + i] =
+          (((*sync_buffer_)[0][start_ix + i] * muting_window) +
+           ((*output)[0][i] * unmuting_window) + 16384) >>
+          15;
+      muting_window += muting_window_increment;
+      unmuting_window += unmuting_window_increment;
+    }
+    // Remove `overlap_length_` samples from the front of `output` since they
+    // were mixed into `sync_buffer_` above.
+    output->PopFront(overlap_length_);
+  }
+  first_call_ = false;
+  return kOK;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.h b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.h
new file mode 100644
index 0000000000..31fcee31d0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise.h
@@ -0,0 +1,72 @@
+/*
+ *  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_NETEQ_COMFORT_NOISE_H_
+#define MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
+
+#include <stddef.h>
+
+namespace webrtc {
+
+// Forward declarations.
+class AudioMultiVector;
+class DecoderDatabase;
+class SyncBuffer;
+struct Packet;
+
+// This class acts as an interface to the CNG generator.
+class ComfortNoise {
+ public:
+  enum ReturnCodes {
+    kOK = 0,
+    kUnknownPayloadType,
+    kInternalError,
+    kMultiChannelNotSupported
+  };
+
+  ComfortNoise(int fs_hz,
+               DecoderDatabase* decoder_database,
+               SyncBuffer* sync_buffer)
+      : fs_hz_(fs_hz),
+        first_call_(true),
+        overlap_length_(5 * fs_hz_ / 8000),
+        decoder_database_(decoder_database),
+        sync_buffer_(sync_buffer) {}
+
+  ComfortNoise(const ComfortNoise&) = delete;
+  ComfortNoise& operator=(const ComfortNoise&) = delete;
+
+  // Resets the state. Should be called before each new comfort noise period.
+  void Reset();
+
+  // Update the comfort noise generator with the parameters in `packet`.
+  int UpdateParameters(const Packet& packet);
+
+  // Generates `requested_length` samples of comfort noise and writes to
+  // `output`. If this is the first in call after Reset (or first after creating
+  // the object), it will also mix in comfort noise at the end of the
+  // SyncBuffer object provided in the constructor.
+  int Generate(size_t requested_length, AudioMultiVector* output);
+
+  // Returns the last error code that was produced by the comfort noise
+  // decoder. Returns 0 if no error has been encountered since the last reset.
+  int internal_error_code() { return internal_error_code_; }
+
+ private:
+  int fs_hz_;
+  bool first_call_;
+  size_t overlap_length_;
+  DecoderDatabase* decoder_database_;
+  SyncBuffer* sync_buffer_;
+  int internal_error_code_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc
new file mode 100644
index 0000000000..b436800061
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc
@@ -0,0 +1,31 @@
+/*
+ *  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.
+ */
+
+// Unit tests for ComfortNoise class.
+
+#include "modules/audio_coding/neteq/comfort_noise.h"
+
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(ComfortNoise, CreateAndDestroy) {
+  int fs = 8000;
+  MockDecoderDatabase db;
+  SyncBuffer sync_buffer(1, 1000);
+  ComfortNoise cn(fs, &db, &sync_buffer);
+  EXPECT_CALL(db, Die());  // Called when `db` goes out of scope.
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.cc b/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.cc
new file mode 100644
index 0000000000..37ed9374f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.cc
@@ -0,0 +1,55 @@
+/*
+ *  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/neteq/cross_correlation.h"
+
+#include <cstdlib>
+#include <limits>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+// This function decides the overflow-protecting scaling and calls
+// WebRtcSpl_CrossCorrelation.
+int CrossCorrelationWithAutoShift(const int16_t* sequence_1,
+                                  const int16_t* sequence_2,
+                                  size_t sequence_1_length,
+                                  size_t cross_correlation_length,
+                                  int cross_correlation_step,
+                                  int32_t* cross_correlation) {
+  // Find the element that has the maximum absolute value of sequence_1 and 2.
+  // Note that these values may be negative.
+  const int16_t max_1 =
+      WebRtcSpl_MaxAbsElementW16(sequence_1, sequence_1_length);
+  const int sequence_2_shift =
+      cross_correlation_step * (static_cast<int>(cross_correlation_length) - 1);
+  const int16_t* sequence_2_start =
+      sequence_2_shift >= 0 ? sequence_2 : sequence_2 + sequence_2_shift;
+  const size_t sequence_2_length =
+      sequence_1_length + std::abs(sequence_2_shift);
+  const int16_t max_2 =
+      WebRtcSpl_MaxAbsElementW16(sequence_2_start, sequence_2_length);
+
+  // In order to avoid overflow when computing the sum we should scale the
+  // samples so that (in_vector_length * max_1 * max_2) will not overflow.
+  const int64_t max_value =
+      abs(max_1 * max_2) * static_cast<int64_t>(sequence_1_length);
+  const int32_t factor = max_value >> 31;
+  const int scaling = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+
+  WebRtcSpl_CrossCorrelation(cross_correlation, sequence_1, sequence_2,
+                             sequence_1_length, cross_correlation_length,
+                             scaling, cross_correlation_step);
+
+  return scaling;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.h b/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.h
new file mode 100644
index 0000000000..5082ce6a30
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/cross_correlation.h
@@ -0,0 +1,51 @@
+/*
+ *  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_NETEQ_CROSS_CORRELATION_H_
+#define MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// The function calculates the cross-correlation between two sequences
+// `sequence_1` and `sequence_2`. `sequence_1` is taken as reference, with
+// `sequence_1_length` as its length. `sequence_2` slides for the calculation of
+// cross-correlation. The result will be saved in `cross_correlation`.
+// `cross_correlation_length` correlation points are calculated.
+// The corresponding lag starts from 0, and increases with a step of
+// `cross_correlation_step`. The result is without normalization. To avoid
+// overflow, the result will be right shifted. The amount of shifts will be
+// returned.
+//
+// Input:
+//     - sequence_1     : First sequence (reference).
+//     - sequence_2     : Second sequence (sliding during calculation).
+//     - sequence_1_length : Length of `sequence_1`.
+//     - cross_correlation_length : Number of cross-correlations to calculate.
+//     - cross_correlation_step : Step in the lag for the cross-correlation.
+//
+// Output:
+//      - cross_correlation : The cross-correlation in Q(-right_shifts)
+//
+// Return:
+//      Number of right shifts in cross_correlation.
+
+int CrossCorrelationWithAutoShift(const int16_t* sequence_1,
+                                  const int16_t* sequence_2,
+                                  size_t sequence_1_length,
+                                  size_t cross_correlation_length,
+                                  int cross_correlation_step,
+                                  int32_t* cross_correlation);
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
new file mode 100644
index 0000000000..91b0252d2c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
@@ -0,0 +1,515 @@
+/*
+ *  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 "modules/audio_coding/neteq/decision_logic.h"
+
+#include <stdio.h>
+
+#include <cstdint>
+#include <memory>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "modules/audio_coding/neteq/packet_arrival_history.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kPostponeDecodingLevel = 50;
+constexpr int kTargetLevelWindowMs = 100;
+constexpr int kMaxWaitForPacketTicks = 10;
+// The granularity of delay adjustments (accelerate/preemptive expand) is 15ms,
+// but round up since the clock has a granularity of 10ms.
+constexpr int kDelayAdjustmentGranularityMs = 20;
+
+std::unique_ptr<DelayManager> CreateDelayManager(
+    const NetEqController::Config& neteq_config) {
+  DelayManager::Config config;
+  config.max_packets_in_buffer = neteq_config.max_packets_in_buffer;
+  config.base_minimum_delay_ms = neteq_config.base_min_delay_ms;
+  config.Log();
+  return std::make_unique<DelayManager>(config, neteq_config.tick_timer);
+}
+
+bool IsTimestretch(NetEq::Mode mode) {
+  return mode == NetEq::Mode::kAccelerateSuccess ||
+         mode == NetEq::Mode::kAccelerateLowEnergy ||
+         mode == NetEq::Mode::kPreemptiveExpandSuccess ||
+         mode == NetEq::Mode::kPreemptiveExpandLowEnergy;
+}
+
+bool IsCng(NetEq::Mode mode) {
+  return mode == NetEq::Mode::kRfc3389Cng ||
+         mode == NetEq::Mode::kCodecInternalCng;
+}
+
+bool IsExpand(NetEq::Mode mode) {
+  return mode == NetEq::Mode::kExpand || mode == NetEq::Mode::kCodecPlc;
+}
+
+}  // namespace
+
+DecisionLogic::Config::Config() {
+  StructParametersParser::Create(
+      "enable_stable_playout_delay", &enable_stable_playout_delay,  //
+      "reinit_after_expands", &reinit_after_expands,                //
+      "packet_history_size_ms", &packet_history_size_ms,            //
+      "cng_timeout_ms", &cng_timeout_ms,                            //
+      "deceleration_target_level_offset_ms",
+      &deceleration_target_level_offset_ms)
+      ->Parse(webrtc::field_trial::FindFullName(
+          "WebRTC-Audio-NetEqDecisionLogicConfig"));
+  RTC_LOG(LS_INFO) << "NetEq decision logic config:"
+                   << " enable_stable_playout_delay="
+                   << enable_stable_playout_delay
+                   << " reinit_after_expands=" << reinit_after_expands
+                   << " packet_history_size_ms=" << packet_history_size_ms
+                   << " cng_timeout_ms=" << cng_timeout_ms.value_or(-1)
+                   << " deceleration_target_level_offset_ms="
+                   << deceleration_target_level_offset_ms;
+}
+
+DecisionLogic::DecisionLogic(NetEqController::Config config)
+    : DecisionLogic(config,
+                    CreateDelayManager(config),
+                    std::make_unique<BufferLevelFilter>()) {}
+
+DecisionLogic::DecisionLogic(
+    NetEqController::Config config,
+    std::unique_ptr<DelayManager> delay_manager,
+    std::unique_ptr<BufferLevelFilter> buffer_level_filter)
+    : delay_manager_(std::move(delay_manager)),
+      buffer_level_filter_(std::move(buffer_level_filter)),
+      packet_arrival_history_(config_.packet_history_size_ms),
+      tick_timer_(config.tick_timer),
+      disallow_time_stretching_(!config.allow_time_stretching),
+      timescale_countdown_(
+          tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1)) {}
+
+DecisionLogic::~DecisionLogic() = default;
+
+void DecisionLogic::SoftReset() {
+  packet_length_samples_ = 0;
+  sample_memory_ = 0;
+  prev_time_scale_ = false;
+  timescale_countdown_ =
+      tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1);
+  time_stretched_cn_samples_ = 0;
+  delay_manager_->Reset();
+  buffer_level_filter_->Reset();
+  packet_arrival_history_.Reset();
+  last_playout_delay_ms_ = 0;
+}
+
+void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
+  // TODO(hlundin): Change to an enumerator and skip assert.
+  RTC_DCHECK(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 ||
+             fs_hz == 48000);
+  sample_rate_khz_ = fs_hz / 1000;
+  output_size_samples_ = output_size_samples;
+  packet_arrival_history_.set_sample_rate(fs_hz);
+}
+
+NetEq::Operation DecisionLogic::GetDecision(const NetEqStatus& status,
+                                            bool* reset_decoder) {
+  // If last mode was CNG (or Expand, since this could be covering up for
+  // a lost CNG packet), remember that CNG is on. This is needed if comfort
+  // noise is interrupted by DTMF.
+  if (status.last_mode == NetEq::Mode::kRfc3389Cng) {
+    cng_state_ = kCngRfc3389On;
+  } else if (status.last_mode == NetEq::Mode::kCodecInternalCng) {
+    cng_state_ = kCngInternalOn;
+  }
+
+  if (IsExpand(status.last_mode)) {
+    ++num_consecutive_expands_;
+  } else {
+    num_consecutive_expands_ = 0;
+  }
+
+  if (!IsExpand(status.last_mode) && !IsCng(status.last_mode)) {
+    last_playout_delay_ms_ = GetPlayoutDelayMs(status);
+  }
+
+  prev_time_scale_ = prev_time_scale_ && IsTimestretch(status.last_mode);
+  if (prev_time_scale_) {
+    timescale_countdown_ = tick_timer_->GetNewCountdown(kMinTimescaleInterval);
+  }
+  if (!IsCng(status.last_mode)) {
+    FilterBufferLevel(status.packet_buffer_info.span_samples);
+  }
+
+  // Guard for errors, to avoid getting stuck in error mode.
+  if (status.last_mode == NetEq::Mode::kError) {
+    if (!status.next_packet) {
+      return NetEq::Operation::kExpand;
+    } else {
+      // Use kUndefined to flag for a reset.
+      return NetEq::Operation::kUndefined;
+    }
+  }
+
+  if (status.next_packet && status.next_packet->is_cng) {
+    return CngOperation(status);
+  }
+
+  // Handle the case with no packet at all available (except maybe DTMF).
+  if (!status.next_packet) {
+    return NoPacket(status);
+  }
+
+  // If the expand period was very long, reset NetEQ since it is likely that the
+  // sender was restarted.
+  if (num_consecutive_expands_ > config_.reinit_after_expands) {
+    *reset_decoder = true;
+    return NetEq::Operation::kNormal;
+  }
+
+  // Make sure we don't restart audio too soon after an expansion to avoid
+  // running out of data right away again. We should only wait if there are no
+  // DTX or CNG packets in the buffer (otherwise we should just play out what we
+  // have, since we cannot know the exact duration of DTX or CNG packets), and
+  // if the mute factor is low enough (otherwise the expansion was short enough
+  // to not be noticable).
+  // Note that the MuteFactor is in Q14, so a value of 16384 corresponds to 1.
+  const int target_level_samples = TargetLevelMs() * sample_rate_khz_;
+  if (!config_.enable_stable_playout_delay && IsExpand(status.last_mode) &&
+      status.expand_mutefactor < 16384 / 2 &&
+      status.packet_buffer_info.span_samples <
+          static_cast<size_t>(target_level_samples * kPostponeDecodingLevel /
+                              100) &&
+      !status.packet_buffer_info.dtx_or_cng) {
+    return NetEq::Operation::kExpand;
+  }
+
+  const uint32_t five_seconds_samples =
+      static_cast<uint32_t>(5000 * sample_rate_khz_);
+  // Check if the required packet is available.
+  if (status.target_timestamp == status.next_packet->timestamp) {
+    return ExpectedPacketAvailable(status);
+  }
+  if (!PacketBuffer::IsObsoleteTimestamp(status.next_packet->timestamp,
+                                         status.target_timestamp,
+                                         five_seconds_samples)) {
+    return FuturePacketAvailable(status);
+  }
+  // This implies that available_timestamp < target_timestamp, which can
+  // happen when a new stream or codec is received. Signal for a reset.
+  return NetEq::Operation::kUndefined;
+}
+
+void DecisionLogic::NotifyMutedState() {
+  ++num_consecutive_expands_;
+}
+
+int DecisionLogic::TargetLevelMs() const {
+  int target_delay_ms = delay_manager_->TargetDelayMs();
+  if (!config_.enable_stable_playout_delay) {
+    target_delay_ms =
+        std::max(target_delay_ms,
+                 static_cast<int>(packet_length_samples_ / sample_rate_khz_));
+  }
+  return target_delay_ms;
+}
+
+int DecisionLogic::UnlimitedTargetLevelMs() const {
+  return delay_manager_->UnlimitedTargetLevelMs();
+}
+
+int DecisionLogic::GetFilteredBufferLevel() const {
+  if (config_.enable_stable_playout_delay) {
+    return last_playout_delay_ms_ * sample_rate_khz_;
+  }
+  return buffer_level_filter_->filtered_current_level();
+}
+
+absl::optional<int> DecisionLogic::PacketArrived(
+    int fs_hz,
+    bool should_update_stats,
+    const PacketArrivedInfo& info) {
+  buffer_flush_ = buffer_flush_ || info.buffer_flush;
+  if (!should_update_stats || info.is_cng_or_dtmf) {
+    return absl::nullopt;
+  }
+  if (info.packet_length_samples > 0 && fs_hz > 0 &&
+      info.packet_length_samples != packet_length_samples_) {
+    packet_length_samples_ = info.packet_length_samples;
+    delay_manager_->SetPacketAudioLength(packet_length_samples_ * 1000 / fs_hz);
+  }
+  int64_t time_now_ms = tick_timer_->ticks() * tick_timer_->ms_per_tick();
+  packet_arrival_history_.Insert(info.main_timestamp, time_now_ms);
+  if (packet_arrival_history_.size() < 2) {
+    // No meaningful delay estimate unless at least 2 packets have arrived.
+    return absl::nullopt;
+  }
+  int arrival_delay_ms =
+      packet_arrival_history_.GetDelayMs(info.main_timestamp, time_now_ms);
+  bool reordered =
+      !packet_arrival_history_.IsNewestRtpTimestamp(info.main_timestamp);
+  delay_manager_->Update(arrival_delay_ms, reordered);
+  return arrival_delay_ms;
+}
+
+void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples) {
+  buffer_level_filter_->SetTargetBufferLevel(TargetLevelMs());
+
+  int time_stretched_samples = time_stretched_cn_samples_;
+  if (prev_time_scale_) {
+    time_stretched_samples += sample_memory_;
+  }
+
+  if (buffer_flush_) {
+    buffer_level_filter_->SetFilteredBufferLevel(buffer_size_samples);
+    buffer_flush_ = false;
+  } else {
+    buffer_level_filter_->Update(buffer_size_samples, time_stretched_samples);
+  }
+  prev_time_scale_ = false;
+  time_stretched_cn_samples_ = 0;
+}
+
+NetEq::Operation DecisionLogic::CngOperation(
+    NetEqController::NetEqStatus status) {
+  // Signed difference between target and available timestamp.
+  int32_t timestamp_diff = static_cast<int32_t>(
+      static_cast<uint32_t>(status.generated_noise_samples +
+                            status.target_timestamp) -
+      status.next_packet->timestamp);
+  int optimal_level_samp = TargetLevelMs() * sample_rate_khz_;
+  const int64_t excess_waiting_time_samp =
+      -static_cast<int64_t>(timestamp_diff) - optimal_level_samp;
+
+  if (excess_waiting_time_samp > optimal_level_samp / 2) {
+    // The waiting time for this packet will be longer than 1.5
+    // times the wanted buffer delay. Apply fast-forward to cut the
+    // waiting time down to the optimal.
+    noise_fast_forward_ = rtc::saturated_cast<size_t>(noise_fast_forward_ +
+                                                      excess_waiting_time_samp);
+    timestamp_diff =
+        rtc::saturated_cast<int32_t>(timestamp_diff + excess_waiting_time_samp);
+  }
+
+  if (timestamp_diff < 0 && status.last_mode == NetEq::Mode::kRfc3389Cng) {
+    // Not time to play this packet yet. Wait another round before using this
+    // packet. Keep on playing CNG from previous CNG parameters.
+    return NetEq::Operation::kRfc3389CngNoPacket;
+  } else {
+    // Otherwise, go for the CNG packet now.
+    noise_fast_forward_ = 0;
+    return NetEq::Operation::kRfc3389Cng;
+  }
+}
+
+NetEq::Operation DecisionLogic::NoPacket(NetEqController::NetEqStatus status) {
+  if (cng_state_ == kCngRfc3389On) {
+    // Keep on playing comfort noise.
+    return NetEq::Operation::kRfc3389CngNoPacket;
+  } else if (cng_state_ == kCngInternalOn) {
+    // Stop CNG after a timeout.
+    if (config_.cng_timeout_ms &&
+        status.generated_noise_samples >
+            static_cast<size_t>(*config_.cng_timeout_ms * sample_rate_khz_)) {
+      return NetEq::Operation::kExpand;
+    }
+    return NetEq::Operation::kCodecInternalCng;
+  } else if (status.play_dtmf) {
+    return NetEq::Operation::kDtmf;
+  } else {
+    // Nothing to play, do expand.
+    return NetEq::Operation::kExpand;
+  }
+}
+
+NetEq::Operation DecisionLogic::ExpectedPacketAvailable(
+    NetEqController::NetEqStatus status) {
+  if (!disallow_time_stretching_ && status.last_mode != NetEq::Mode::kExpand &&
+      !status.play_dtmf) {
+    if (config_.enable_stable_playout_delay) {
+      const int playout_delay_ms = GetPlayoutDelayMs(status);
+      if (playout_delay_ms >= HighThreshold() << 2) {
+        return NetEq::Operation::kFastAccelerate;
+      }
+      if (TimescaleAllowed()) {
+        if (playout_delay_ms >= HighThreshold()) {
+          return NetEq::Operation::kAccelerate;
+        }
+        if (playout_delay_ms < LowThreshold()) {
+          return NetEq::Operation::kPreemptiveExpand;
+        }
+      }
+    } else {
+      const int target_level_samples = TargetLevelMs() * sample_rate_khz_;
+      const int low_limit = std::max(
+          target_level_samples * 3 / 4,
+          target_level_samples -
+              config_.deceleration_target_level_offset_ms * sample_rate_khz_);
+      const int high_limit = std::max(
+          target_level_samples,
+          low_limit + kDelayAdjustmentGranularityMs * sample_rate_khz_);
+
+      const int buffer_level_samples =
+          buffer_level_filter_->filtered_current_level();
+      if (buffer_level_samples >= high_limit << 2)
+        return NetEq::Operation::kFastAccelerate;
+      if (TimescaleAllowed()) {
+        if (buffer_level_samples >= high_limit)
+          return NetEq::Operation::kAccelerate;
+        if (buffer_level_samples < low_limit)
+          return NetEq::Operation::kPreemptiveExpand;
+      }
+    }
+  }
+  return NetEq::Operation::kNormal;
+}
+
+NetEq::Operation DecisionLogic::FuturePacketAvailable(
+    NetEqController::NetEqStatus status) {
+  // Required packet is not available, but a future packet is.
+  // Check if we should continue with an ongoing expand because the new packet
+  // is too far into the future.
+  if (IsExpand(status.last_mode) && ShouldContinueExpand(status)) {
+    if (status.play_dtmf) {
+      // Still have DTMF to play, so do not do expand.
+      return NetEq::Operation::kDtmf;
+    } else {
+      // Nothing to play.
+      return NetEq::Operation::kExpand;
+    }
+  }
+
+  if (status.last_mode == NetEq::Mode::kCodecPlc) {
+    return NetEq::Operation::kNormal;
+  }
+
+  // If previous was comfort noise, then no merge is needed.
+  if (IsCng(status.last_mode)) {
+    uint32_t timestamp_leap =
+        status.next_packet->timestamp - status.target_timestamp;
+    const bool generated_enough_noise =
+        status.generated_noise_samples >= timestamp_leap;
+
+    int playout_delay_ms = GetNextPacketDelayMs(status);
+    const bool above_target_delay = playout_delay_ms > HighThresholdCng();
+    const bool below_target_delay = playout_delay_ms < LowThresholdCng();
+    // Keep the delay same as before CNG, but make sure that it is within the
+    // target window.
+    if ((generated_enough_noise && !below_target_delay) || above_target_delay) {
+      time_stretched_cn_samples_ =
+          timestamp_leap - status.generated_noise_samples;
+      return NetEq::Operation::kNormal;
+    }
+
+    if (status.last_mode == NetEq::Mode::kRfc3389Cng) {
+      return NetEq::Operation::kRfc3389CngNoPacket;
+    }
+    return NetEq::Operation::kCodecInternalCng;
+  }
+
+  // Do not merge unless we have done an expand before.
+  if (status.last_mode == NetEq::Mode::kExpand) {
+    return NetEq::Operation::kMerge;
+  } else if (status.play_dtmf) {
+    // Play DTMF instead of expand.
+    return NetEq::Operation::kDtmf;
+  } else {
+    return NetEq::Operation::kExpand;
+  }
+}
+
+bool DecisionLogic::UnderTargetLevel() const {
+  return buffer_level_filter_->filtered_current_level() <
+         TargetLevelMs() * sample_rate_khz_;
+}
+
+bool DecisionLogic::ReinitAfterExpands(uint32_t timestamp_leap) const {
+  return timestamp_leap >= static_cast<uint32_t>(output_size_samples_ *
+                                                 config_.reinit_after_expands);
+}
+
+bool DecisionLogic::PacketTooEarly(uint32_t timestamp_leap) const {
+  return timestamp_leap >
+         static_cast<uint32_t>(output_size_samples_ * num_consecutive_expands_);
+}
+
+bool DecisionLogic::MaxWaitForPacket() const {
+  return num_consecutive_expands_ >= kMaxWaitForPacketTicks;
+}
+
+bool DecisionLogic::ShouldContinueExpand(
+    NetEqController::NetEqStatus status) const {
+  uint32_t timestamp_leap =
+      status.next_packet->timestamp - status.target_timestamp;
+  if (config_.enable_stable_playout_delay) {
+    return GetNextPacketDelayMs(status) < HighThreshold() &&
+           PacketTooEarly(timestamp_leap);
+  }
+  return !ReinitAfterExpands(timestamp_leap) && !MaxWaitForPacket() &&
+         PacketTooEarly(timestamp_leap) && UnderTargetLevel();
+}
+
+int DecisionLogic::GetNextPacketDelayMs(
+    NetEqController::NetEqStatus status) const {
+  if (config_.enable_stable_playout_delay) {
+    return packet_arrival_history_.GetDelayMs(
+        status.next_packet->timestamp,
+        tick_timer_->ticks() * tick_timer_->ms_per_tick());
+  }
+  return status.packet_buffer_info.span_samples / sample_rate_khz_;
+}
+
+int DecisionLogic::GetPlayoutDelayMs(
+    NetEqController::NetEqStatus status) const {
+  uint32_t playout_timestamp =
+      status.target_timestamp - status.sync_buffer_samples;
+  return packet_arrival_history_.GetDelayMs(
+      playout_timestamp, tick_timer_->ticks() * tick_timer_->ms_per_tick());
+}
+
+int DecisionLogic::LowThreshold() const {
+  int target_delay_ms = TargetLevelMs();
+  return std::max(
+      target_delay_ms * 3 / 4,
+      target_delay_ms - config_.deceleration_target_level_offset_ms);
+}
+
+int DecisionLogic::HighThreshold() const {
+  if (config_.enable_stable_playout_delay) {
+    return std::max(TargetLevelMs(), packet_arrival_history_.GetMaxDelayMs()) +
+           kDelayAdjustmentGranularityMs;
+  }
+  return std::max(TargetLevelMs(),
+                  LowThreshold() + kDelayAdjustmentGranularityMs);
+}
+
+int DecisionLogic::LowThresholdCng() const {
+  if (config_.enable_stable_playout_delay) {
+    return LowThreshold();
+  }
+  return std::max(0, TargetLevelMs() - kTargetLevelWindowMs / 2);
+}
+
+int DecisionLogic::HighThresholdCng() const {
+  if (config_.enable_stable_playout_delay) {
+    return HighThreshold();
+  }
+  return TargetLevelMs() + kTargetLevelWindowMs / 2;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.h b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.h
new file mode 100644
index 0000000000..8d1ff4d622
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.h
@@ -0,0 +1,202 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
+#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
+
+#include <memory>
+
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/packet_arrival_history.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+// This is the class for the decision tree implementation.
+class DecisionLogic : public NetEqController {
+ public:
+  DecisionLogic(NetEqController::Config config);
+  DecisionLogic(NetEqController::Config config,
+                std::unique_ptr<DelayManager> delay_manager,
+                std::unique_ptr<BufferLevelFilter> buffer_level_filter);
+
+  ~DecisionLogic() override;
+
+  DecisionLogic(const DecisionLogic&) = delete;
+  DecisionLogic& operator=(const DecisionLogic&) = delete;
+
+  // Not used.
+  void Reset() override {}
+
+  // Resets parts of the state. Typically done when switching codecs.
+  void SoftReset() override;
+
+  // Sets the sample rate and the output block size.
+  void SetSampleRate(int fs_hz, size_t output_size_samples) override;
+
+  // Given info about the latest received packet, and current jitter buffer
+  // status, returns the operation. `target_timestamp` and `expand_mutefactor`
+  // are provided for reference. `last_packet_samples` is the number of samples
+  // obtained from the last decoded frame. If there is a packet available, it
+  // should be supplied in `packet`; otherwise it should be NULL. The mode
+  // resulting from the last call to NetEqImpl::GetAudio is supplied in
+  // `last_mode`. If there is a DTMF event to play, `play_dtmf` should be set to
+  // true. The output variable `reset_decoder` will be set to true if a reset is
+  // required; otherwise it is left unchanged (i.e., it can remain true if it
+  // was true before the call).
+  NetEq::Operation GetDecision(const NetEqController::NetEqStatus& status,
+                               bool* reset_decoder) override;
+
+  // These methods test the `cng_state_` for different conditions.
+  bool CngRfc3389On() const override { return cng_state_ == kCngRfc3389On; }
+  bool CngOff() const override { return cng_state_ == kCngOff; }
+
+  // Resets the `cng_state_` to kCngOff.
+  void SetCngOff() override { cng_state_ = kCngOff; }
+
+  void ExpandDecision(NetEq::Operation operation) override {}
+
+  // Adds `value` to `sample_memory_`.
+  void AddSampleMemory(int32_t value) override { sample_memory_ += value; }
+
+  int TargetLevelMs() const override;
+
+  int UnlimitedTargetLevelMs() const override;
+
+  absl::optional<int> PacketArrived(int fs_hz,
+                                    bool should_update_stats,
+                                    const PacketArrivedInfo& info) override;
+
+  void RegisterEmptyPacket() override {}
+
+  void NotifyMutedState() override;
+
+  bool SetMaximumDelay(int delay_ms) override {
+    return delay_manager_->SetMaximumDelay(delay_ms);
+  }
+  bool SetMinimumDelay(int delay_ms) override {
+    return delay_manager_->SetMinimumDelay(delay_ms);
+  }
+  bool SetBaseMinimumDelay(int delay_ms) override {
+    return delay_manager_->SetBaseMinimumDelay(delay_ms);
+  }
+  int GetBaseMinimumDelay() const override {
+    return delay_manager_->GetBaseMinimumDelay();
+  }
+  bool PeakFound() const override { return false; }
+
+  int GetFilteredBufferLevel() const override;
+
+  // Accessors and mutators.
+  void set_sample_memory(int32_t value) override { sample_memory_ = value; }
+  size_t noise_fast_forward() const override { return noise_fast_forward_; }
+  size_t packet_length_samples() const override {
+    return packet_length_samples_;
+  }
+  void set_packet_length_samples(size_t value) override {
+    packet_length_samples_ = value;
+  }
+  void set_prev_time_scale(bool value) override { prev_time_scale_ = value; }
+
+ private:
+  // The value 5 sets maximum time-stretch rate to about 100 ms/s.
+  static const int kMinTimescaleInterval = 5;
+
+  enum CngState { kCngOff, kCngRfc3389On, kCngInternalOn };
+
+  // Updates the `buffer_level_filter_` with the current buffer level
+  // `buffer_size_samples`.
+  void FilterBufferLevel(size_t buffer_size_samples);
+
+  // Returns the operation given that the next available packet is a comfort
+  // noise payload (RFC 3389 only, not codec-internal).
+  virtual NetEq::Operation CngOperation(NetEqController::NetEqStatus status);
+
+  // Returns the operation given that no packets are available (except maybe
+  // a DTMF event, flagged by setting `play_dtmf` true).
+  virtual NetEq::Operation NoPacket(NetEqController::NetEqStatus status);
+
+  // Returns the operation to do given that the expected packet is available.
+  virtual NetEq::Operation ExpectedPacketAvailable(
+      NetEqController::NetEqStatus status);
+
+  // Returns the operation to do given that the expected packet is not
+  // available, but a packet further into the future is at hand.
+  virtual NetEq::Operation FuturePacketAvailable(
+      NetEqController::NetEqStatus status);
+
+  // Checks if enough time has elapsed since the last successful timescale
+  // operation was done (i.e., accelerate or preemptive expand).
+  bool TimescaleAllowed() const {
+    return !timescale_countdown_ || timescale_countdown_->Finished();
+  }
+
+  // Checks if the current (filtered) buffer level is under the target level.
+  bool UnderTargetLevel() const;
+
+  // Checks if `timestamp_leap` is so long into the future that a reset due
+  // to exceeding kReinitAfterExpands will be done.
+  bool ReinitAfterExpands(uint32_t timestamp_leap) const;
+
+  // Checks if we still have not done enough expands to cover the distance from
+  // the last decoded packet to the next available packet, the distance beeing
+  // conveyed in `timestamp_leap`.
+  bool PacketTooEarly(uint32_t timestamp_leap) const;
+
+  bool MaxWaitForPacket() const;
+
+  bool ShouldContinueExpand(NetEqController::NetEqStatus status) const;
+
+  int GetNextPacketDelayMs(NetEqController::NetEqStatus status) const;
+  int GetPlayoutDelayMs(NetEqController::NetEqStatus status) const;
+
+  int LowThreshold() const;
+  int HighThreshold() const;
+  int LowThresholdCng() const;
+  int HighThresholdCng() const;
+
+  // Runtime configurable options through field trial
+  // WebRTC-Audio-NetEqDecisionLogicConfig.
+  struct Config {
+    Config();
+
+    bool enable_stable_playout_delay = false;
+    int reinit_after_expands = 100;
+    int deceleration_target_level_offset_ms = 85;
+    int packet_history_size_ms = 2000;
+    absl::optional<int> cng_timeout_ms;
+  };
+  Config config_;
+  std::unique_ptr<DelayManager> delay_manager_;
+  std::unique_ptr<BufferLevelFilter> buffer_level_filter_;
+  PacketArrivalHistory packet_arrival_history_;
+  const TickTimer* tick_timer_;
+  int sample_rate_khz_;
+  size_t output_size_samples_;
+  CngState cng_state_ = kCngOff;  // Remember if comfort noise is interrupted by
+                                  // other event (e.g., DTMF).
+  size_t noise_fast_forward_ = 0;
+  size_t packet_length_samples_ = 0;
+  int sample_memory_ = 0;
+  bool prev_time_scale_ = false;
+  bool disallow_time_stretching_;
+  std::unique_ptr<TickTimer::Countdown> timescale_countdown_;
+  int num_consecutive_expands_ = 0;
+  int time_stretched_cn_samples_ = 0;
+  bool buffer_flush_ = false;
+  int last_playout_delay_ms_ = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
new file mode 100644
index 0000000000..6150c9a6db
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -0,0 +1,219 @@
+/*
+ *  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.
+ */
+
+// Unit tests for DecisionLogic class and derived classes.
+
+#include "modules/audio_coding/neteq/decision_logic.h"
+
+#include "api/neteq/neteq_controller.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/mock/mock_buffer_level_filter.h"
+#include "modules/audio_coding/neteq/mock/mock_delay_manager.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kSampleRate = 8000;
+constexpr int kSamplesPerMs = kSampleRate / 1000;
+constexpr int kOutputSizeSamples = kSamplesPerMs * 10;
+constexpr int kMinTimescaleInterval = 5;
+
+NetEqController::NetEqStatus CreateNetEqStatus(NetEq::Mode last_mode,
+                                               int current_delay_ms) {
+  NetEqController::NetEqStatus status;
+  status.play_dtmf = false;
+  status.last_mode = last_mode;
+  status.target_timestamp = 1234;
+  status.generated_noise_samples = 0;
+  status.expand_mutefactor = 0;
+  status.packet_buffer_info.num_samples = current_delay_ms * kSamplesPerMs;
+  status.packet_buffer_info.span_samples = current_delay_ms * kSamplesPerMs;
+  status.packet_buffer_info.span_samples_no_dtx =
+      current_delay_ms * kSamplesPerMs;
+  status.packet_buffer_info.dtx_or_cng = false;
+  status.next_packet = {status.target_timestamp, false, false};
+  return status;
+}
+
+using ::testing::Return;
+
+}  // namespace
+
+class DecisionLogicTest : public ::testing::Test {
+ protected:
+  DecisionLogicTest() {
+    test::ScopedFieldTrials trials(
+        "WebRTC-Audio-NetEqDecisionLogicConfig/cng_timeout_ms:1000/");
+    NetEqController::Config config;
+    config.tick_timer = &tick_timer_;
+    config.allow_time_stretching = true;
+    auto delay_manager = std::make_unique<MockDelayManager>(
+        DelayManager::Config(), config.tick_timer);
+    mock_delay_manager_ = delay_manager.get();
+    auto buffer_level_filter = std::make_unique<MockBufferLevelFilter>();
+    mock_buffer_level_filter_ = buffer_level_filter.get();
+    decision_logic_ = std::make_unique<DecisionLogic>(
+        config, std::move(delay_manager), std::move(buffer_level_filter));
+    decision_logic_->SetSampleRate(kSampleRate, kOutputSizeSamples);
+  }
+
+  TickTimer tick_timer_;
+  std::unique_ptr<DecisionLogic> decision_logic_;
+  MockDelayManager* mock_delay_manager_;
+  MockBufferLevelFilter* mock_buffer_level_filter_;
+};
+
+TEST_F(DecisionLogicTest, NormalOperation) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(100));
+  EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level())
+      .WillRepeatedly(Return(90 * kSamplesPerMs));
+
+  bool reset_decoder = false;
+  tick_timer_.Increment(kMinTimescaleInterval + 1);
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kNormal, 100), &reset_decoder),
+            NetEq::Operation::kNormal);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, Accelerate) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(100));
+  EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level())
+      .WillRepeatedly(Return(110 * kSamplesPerMs));
+
+  bool reset_decoder = false;
+  tick_timer_.Increment(kMinTimescaleInterval + 1);
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kNormal, 100), &reset_decoder),
+            NetEq::Operation::kAccelerate);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, FastAccelerate) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(100));
+  EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level())
+      .WillRepeatedly(Return(400 * kSamplesPerMs));
+
+  bool reset_decoder = false;
+  tick_timer_.Increment(kMinTimescaleInterval + 1);
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kNormal, 100), &reset_decoder),
+            NetEq::Operation::kFastAccelerate);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, PreemptiveExpand) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(100));
+  EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level())
+      .WillRepeatedly(Return(50 * kSamplesPerMs));
+
+  bool reset_decoder = false;
+  tick_timer_.Increment(kMinTimescaleInterval + 1);
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kNormal, 100), &reset_decoder),
+            NetEq::Operation::kPreemptiveExpand);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, DecelerationTargetLevelOffset) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(500));
+  EXPECT_CALL(*mock_buffer_level_filter_, filtered_current_level())
+      .WillRepeatedly(Return(400 * kSamplesPerMs));
+
+  bool reset_decoder = false;
+  tick_timer_.Increment(kMinTimescaleInterval + 1);
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kNormal, 400), &reset_decoder),
+            NetEq::Operation::kPreemptiveExpand);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, PostponeDecodeAfterExpand) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(500));
+
+  // Below 50% target delay threshold.
+  bool reset_decoder = false;
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kExpand, 200), &reset_decoder),
+            NetEq::Operation::kExpand);
+  EXPECT_FALSE(reset_decoder);
+
+  // Above 50% target delay threshold.
+  EXPECT_EQ(decision_logic_->GetDecision(
+                CreateNetEqStatus(NetEq::Mode::kExpand, 250), &reset_decoder),
+            NetEq::Operation::kNormal);
+  EXPECT_FALSE(reset_decoder);
+}
+
+TEST_F(DecisionLogicTest, TimeStrechComfortNoise) {
+  EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
+      .WillRepeatedly(Return(500));
+
+  {
+    bool reset_decoder = false;
+    // Below target window.
+    auto status = CreateNetEqStatus(NetEq::Mode::kCodecInternalCng, 400);
+    status.generated_noise_samples = 400 * kSamplesPerMs;
+    status.next_packet->timestamp =
+        status.target_timestamp + 400 * kSamplesPerMs;
+    EXPECT_EQ(decision_logic_->GetDecision(status, &reset_decoder),
+              NetEq::Operation::kCodecInternalCng);
+    EXPECT_FALSE(reset_decoder);
+  }
+
+  {
+    bool reset_decoder = false;
+    // Above target window.
+    auto status = CreateNetEqStatus(NetEq::Mode::kCodecInternalCng, 600);
+    status.generated_noise_samples = 200 * kSamplesPerMs;
+    status.next_packet->timestamp =
+        status.target_timestamp + 400 * kSamplesPerMs;
+    EXPECT_EQ(decision_logic_->GetDecision(status, &reset_decoder),
+              NetEq::Operation::kNormal);
+    EXPECT_FALSE(reset_decoder);
+
+    // The buffer level filter should be adjusted with the number of samples
+    // that was skipped.
+    int timestamp_leap = status.next_packet->timestamp -
+                         status.target_timestamp -
+                         status.generated_noise_samples;
+    EXPECT_CALL(*mock_buffer_level_filter_,
+                Update(400 * kSamplesPerMs, timestamp_leap));
+    EXPECT_EQ(decision_logic_->GetDecision(
+                  CreateNetEqStatus(NetEq::Mode::kNormal, 400), &reset_decoder),
+              NetEq::Operation::kNormal);
+    EXPECT_FALSE(reset_decoder);
+  }
+}
+
+TEST_F(DecisionLogicTest, CngTimeout) {
+  auto status = CreateNetEqStatus(NetEq::Mode::kCodecInternalCng, 0);
+  status.next_packet = absl::nullopt;
+  status.generated_noise_samples = kSamplesPerMs * 500;
+  bool reset_decoder = false;
+  EXPECT_EQ(decision_logic_->GetDecision(status, &reset_decoder),
+            NetEq::Operation::kCodecInternalCng);
+  status.generated_noise_samples = kSamplesPerMs * 1010;
+  EXPECT_EQ(decision_logic_->GetDecision(status, &reset_decoder),
+            NetEq::Operation::kExpand);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.cc
new file mode 100644
index 0000000000..3447ced1da
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.cc
@@ -0,0 +1,285 @@
+/*
+ *  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/neteq/decoder_database.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <list>
+#include <type_traits>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/audio_format_to_string.h"
+
+namespace webrtc {
+
+DecoderDatabase::DecoderDatabase(
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+    absl::optional<AudioCodecPairId> codec_pair_id)
+    : active_decoder_type_(-1),
+      active_cng_decoder_type_(-1),
+      decoder_factory_(decoder_factory),
+      codec_pair_id_(codec_pair_id) {}
+
+DecoderDatabase::~DecoderDatabase() = default;
+
+DecoderDatabase::DecoderInfo::DecoderInfo(
+    const SdpAudioFormat& audio_format,
+    absl::optional<AudioCodecPairId> codec_pair_id,
+    AudioDecoderFactory* factory,
+    absl::string_view codec_name)
+    : name_(codec_name),
+      audio_format_(audio_format),
+      codec_pair_id_(codec_pair_id),
+      factory_(factory),
+      cng_decoder_(CngDecoder::Create(audio_format)),
+      subtype_(SubtypeFromFormat(audio_format)) {}
+
+DecoderDatabase::DecoderInfo::DecoderInfo(
+    const SdpAudioFormat& audio_format,
+    absl::optional<AudioCodecPairId> codec_pair_id,
+    AudioDecoderFactory* factory)
+    : DecoderInfo(audio_format, codec_pair_id, factory, audio_format.name) {}
+
+DecoderDatabase::DecoderInfo::DecoderInfo(DecoderInfo&&) = default;
+DecoderDatabase::DecoderInfo::~DecoderInfo() = default;
+
+AudioDecoder* DecoderDatabase::DecoderInfo::GetDecoder() const {
+  if (subtype_ != Subtype::kNormal) {
+    // These are handled internally, so they have no AudioDecoder objects.
+    return nullptr;
+  }
+  if (!decoder_) {
+    // TODO(ossu): Keep a check here for now, since a number of tests create
+    // DecoderInfos without factories.
+    RTC_DCHECK(factory_);
+    decoder_ = factory_->MakeAudioDecoder(audio_format_, codec_pair_id_);
+  }
+  RTC_DCHECK(decoder_) << "Failed to create: " << rtc::ToString(audio_format_);
+  return decoder_.get();
+}
+
+bool DecoderDatabase::DecoderInfo::IsType(absl::string_view name) const {
+  return absl::EqualsIgnoreCase(audio_format_.name, name);
+}
+
+absl::optional<DecoderDatabase::DecoderInfo::CngDecoder>
+DecoderDatabase::DecoderInfo::CngDecoder::Create(const SdpAudioFormat& format) {
+  if (absl::EqualsIgnoreCase(format.name, "CN")) {
+    // CN has a 1:1 RTP clock rate to sample rate ratio.
+    const int sample_rate_hz = format.clockrate_hz;
+    RTC_DCHECK(sample_rate_hz == 8000 || sample_rate_hz == 16000 ||
+               sample_rate_hz == 32000 || sample_rate_hz == 48000);
+    return DecoderDatabase::DecoderInfo::CngDecoder{sample_rate_hz};
+  } else {
+    return absl::nullopt;
+  }
+}
+
+DecoderDatabase::DecoderInfo::Subtype
+DecoderDatabase::DecoderInfo::SubtypeFromFormat(const SdpAudioFormat& format) {
+  if (absl::EqualsIgnoreCase(format.name, "CN")) {
+    return Subtype::kComfortNoise;
+  } else if (absl::EqualsIgnoreCase(format.name, "telephone-event")) {
+    return Subtype::kDtmf;
+  } else if (absl::EqualsIgnoreCase(format.name, "red")) {
+    return Subtype::kRed;
+  }
+
+  return Subtype::kNormal;
+}
+
+bool DecoderDatabase::Empty() const {
+  return decoders_.empty();
+}
+
+int DecoderDatabase::Size() const {
+  return static_cast<int>(decoders_.size());
+}
+
+std::vector<int> DecoderDatabase::SetCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  // First collect all payload types that we'll remove or reassign, then remove
+  // them from the database.
+  std::vector<int> changed_payload_types;
+  for (const std::pair<uint8_t, const DecoderInfo&> kv : decoders_) {
+    auto i = codecs.find(kv.first);
+    if (i == codecs.end() || i->second != kv.second.GetFormat()) {
+      changed_payload_types.push_back(kv.first);
+    }
+  }
+  for (int pl_type : changed_payload_types) {
+    Remove(pl_type);
+  }
+
+  // Enter the new and changed payload type mappings into the database.
+  for (const auto& kv : codecs) {
+    const int& rtp_payload_type = kv.first;
+    const SdpAudioFormat& audio_format = kv.second;
+    RTC_DCHECK_GE(rtp_payload_type, 0);
+    RTC_DCHECK_LE(rtp_payload_type, 0x7f);
+    if (decoders_.count(rtp_payload_type) == 0) {
+      decoders_.insert(std::make_pair(
+          rtp_payload_type,
+          DecoderInfo(audio_format, codec_pair_id_, decoder_factory_.get())));
+    } else {
+      // The mapping for this payload type hasn't changed.
+    }
+  }
+
+  return changed_payload_types;
+}
+
+int DecoderDatabase::RegisterPayload(int rtp_payload_type,
+                                     const SdpAudioFormat& audio_format) {
+  if (rtp_payload_type < 0 || rtp_payload_type > 0x7f) {
+    return kInvalidRtpPayloadType;
+  }
+  const auto ret = decoders_.insert(std::make_pair(
+      rtp_payload_type,
+      DecoderInfo(audio_format, codec_pair_id_, decoder_factory_.get())));
+  if (ret.second == false) {
+    // Database already contains a decoder with type `rtp_payload_type`.
+    return kDecoderExists;
+  }
+  return kOK;
+}
+
+int DecoderDatabase::Remove(uint8_t rtp_payload_type) {
+  if (decoders_.erase(rtp_payload_type) == 0) {
+    // No decoder with that `rtp_payload_type`.
+    return kDecoderNotFound;
+  }
+  if (active_decoder_type_ == rtp_payload_type) {
+    active_decoder_type_ = -1;  // No active decoder.
+  }
+  if (active_cng_decoder_type_ == rtp_payload_type) {
+    active_cng_decoder_type_ = -1;  // No active CNG decoder.
+  }
+  return kOK;
+}
+
+void DecoderDatabase::RemoveAll() {
+  decoders_.clear();
+  active_decoder_type_ = -1;      // No active decoder.
+  active_cng_decoder_type_ = -1;  // No active CNG decoder.
+}
+
+const DecoderDatabase::DecoderInfo* DecoderDatabase::GetDecoderInfo(
+    uint8_t rtp_payload_type) const {
+  DecoderMap::const_iterator it = decoders_.find(rtp_payload_type);
+  if (it == decoders_.end()) {
+    // Decoder not found.
+    return NULL;
+  }
+  return &it->second;
+}
+
+int DecoderDatabase::SetActiveDecoder(uint8_t rtp_payload_type,
+                                      bool* new_decoder) {
+  // Check that `rtp_payload_type` exists in the database.
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  if (!info) {
+    // Decoder not found.
+    return kDecoderNotFound;
+  }
+  RTC_CHECK(!info->IsComfortNoise());
+  RTC_DCHECK(new_decoder);
+  *new_decoder = false;
+  if (active_decoder_type_ < 0) {
+    // This is the first active decoder.
+    *new_decoder = true;
+  } else if (active_decoder_type_ != rtp_payload_type) {
+    // Moving from one active decoder to another. Delete the first one.
+    const DecoderInfo* old_info = GetDecoderInfo(active_decoder_type_);
+    RTC_DCHECK(old_info);
+    old_info->DropDecoder();
+    *new_decoder = true;
+  }
+  active_decoder_type_ = rtp_payload_type;
+  return kOK;
+}
+
+AudioDecoder* DecoderDatabase::GetActiveDecoder() const {
+  if (active_decoder_type_ < 0) {
+    // No active decoder.
+    return NULL;
+  }
+  return GetDecoder(active_decoder_type_);
+}
+
+int DecoderDatabase::SetActiveCngDecoder(uint8_t rtp_payload_type) {
+  // Check that `rtp_payload_type` exists in the database.
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  if (!info) {
+    // Decoder not found.
+    return kDecoderNotFound;
+  }
+  if (active_cng_decoder_type_ >= 0 &&
+      active_cng_decoder_type_ != rtp_payload_type) {
+    // Moving from one active CNG decoder to another. Delete the first one.
+    RTC_DCHECK(active_cng_decoder_);
+    active_cng_decoder_.reset();
+  }
+  active_cng_decoder_type_ = rtp_payload_type;
+  return kOK;
+}
+
+ComfortNoiseDecoder* DecoderDatabase::GetActiveCngDecoder() const {
+  if (active_cng_decoder_type_ < 0) {
+    // No active CNG decoder.
+    return NULL;
+  }
+  if (!active_cng_decoder_) {
+    active_cng_decoder_.reset(new ComfortNoiseDecoder);
+  }
+  return active_cng_decoder_.get();
+}
+
+AudioDecoder* DecoderDatabase::GetDecoder(uint8_t rtp_payload_type) const {
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  return info ? info->GetDecoder() : nullptr;
+}
+
+bool DecoderDatabase::IsComfortNoise(uint8_t rtp_payload_type) const {
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  return info && info->IsComfortNoise();
+}
+
+bool DecoderDatabase::IsDtmf(uint8_t rtp_payload_type) const {
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  return info && info->IsDtmf();
+}
+
+bool DecoderDatabase::IsRed(uint8_t rtp_payload_type) const {
+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
+  return info && info->IsRed();
+}
+
+int DecoderDatabase::CheckPayloadTypes(const PacketList& packet_list) const {
+  PacketList::const_iterator it;
+  for (it = packet_list.begin(); it != packet_list.end(); ++it) {
+    if (!GetDecoderInfo(it->payload_type)) {
+      // Payload type is not found.
+      RTC_LOG(LS_WARNING) << "CheckPayloadTypes: unknown RTP payload type "
+                          << static_cast<int>(it->payload_type);
+      return kDecoderNotFound;
+    }
+  }
+  return kOK;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.h b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.h
new file mode 100644
index 0000000000..8cf2019135
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database.h
@@ -0,0 +1,204 @@
+/*
+ *  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_NETEQ_DECODER_DATABASE_H_
+#define MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/scoped_refptr.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "modules/audio_coding/neteq/packet.h"
+
+namespace webrtc {
+
+class DecoderDatabase {
+ public:
+  enum DatabaseReturnCodes {
+    kOK = 0,
+    kInvalidRtpPayloadType = -1,
+    kCodecNotSupported = -2,
+    kInvalidSampleRate = -3,
+    kDecoderExists = -4,
+    kDecoderNotFound = -5,
+    kInvalidPointer = -6
+  };
+
+  // Class that stores decoder info in the database.
+  class DecoderInfo {
+   public:
+    DecoderInfo(const SdpAudioFormat& audio_format,
+                absl::optional<AudioCodecPairId> codec_pair_id,
+                AudioDecoderFactory* factory,
+                absl::string_view codec_name);
+    explicit DecoderInfo(const SdpAudioFormat& audio_format,
+                         absl::optional<AudioCodecPairId> codec_pair_id,
+                         AudioDecoderFactory* factory = nullptr);
+    DecoderInfo(DecoderInfo&&);
+    ~DecoderInfo();
+
+    // Get the AudioDecoder object, creating it first if necessary.
+    AudioDecoder* GetDecoder() const;
+
+    // Delete the AudioDecoder object, unless it's external. (This means we can
+    // always recreate it later if we need it.)
+    void DropDecoder() const { decoder_.reset(); }
+
+    int SampleRateHz() const {
+      if (IsDtmf()) {
+        // DTMF has a 1:1 mapping between clock rate and sample rate.
+        return audio_format_.clockrate_hz;
+      }
+      const AudioDecoder* decoder = GetDecoder();
+      RTC_DCHECK_EQ(1, !!decoder + !!cng_decoder_);
+      return decoder ? decoder->SampleRateHz() : cng_decoder_->sample_rate_hz;
+    }
+
+    const SdpAudioFormat& GetFormat() const { return audio_format_; }
+
+    // Returns true if the decoder's format is comfort noise.
+    bool IsComfortNoise() const {
+      RTC_DCHECK_EQ(!!cng_decoder_, subtype_ == Subtype::kComfortNoise);
+      return subtype_ == Subtype::kComfortNoise;
+    }
+
+    // Returns true if the decoder's format is DTMF.
+    bool IsDtmf() const { return subtype_ == Subtype::kDtmf; }
+
+    // Returns true if the decoder's format is RED.
+    bool IsRed() const { return subtype_ == Subtype::kRed; }
+
+    // Returns true if the decoder's format is named `name`.
+    bool IsType(absl::string_view name) const;
+
+    const std::string& get_name() const { return name_; }
+
+   private:
+    // TODO(ossu): `name_` is kept here while we retain the old external
+    //             decoder interface. Remove this once using an
+    //             AudioDecoderFactory has supplanted the old functionality.
+    const std::string name_;
+
+    const SdpAudioFormat audio_format_;
+    const absl::optional<AudioCodecPairId> codec_pair_id_;
+    AudioDecoderFactory* const factory_;
+    mutable std::unique_ptr<AudioDecoder> decoder_;
+
+    // Set iff this is a comfort noise decoder.
+    struct CngDecoder {
+      static absl::optional<CngDecoder> Create(const SdpAudioFormat& format);
+      int sample_rate_hz;
+    };
+    const absl::optional<CngDecoder> cng_decoder_;
+
+    enum class Subtype : int8_t { kNormal, kComfortNoise, kDtmf, kRed };
+
+    static Subtype SubtypeFromFormat(const SdpAudioFormat& format);
+
+    const Subtype subtype_;
+  };
+
+  // Maximum value for 8 bits, and an invalid RTP payload type (since it is
+  // only 7 bits).
+  static const uint8_t kRtpPayloadTypeError = 0xFF;
+
+  DecoderDatabase(
+      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+      absl::optional<AudioCodecPairId> codec_pair_id);
+
+  virtual ~DecoderDatabase();
+
+  DecoderDatabase(const DecoderDatabase&) = delete;
+  DecoderDatabase& operator=(const DecoderDatabase&) = delete;
+
+  // Returns true if the database is empty.
+  virtual bool Empty() const;
+
+  // Returns the number of decoders registered in the database.
+  virtual int Size() const;
+
+  // Replaces the existing set of decoders with the given set. Returns the
+  // payload types that were reassigned or removed while doing so.
+  virtual std::vector<int> SetCodecs(
+      const std::map<int, SdpAudioFormat>& codecs);
+
+  // Registers a decoder for the given payload type. Returns kOK on success;
+  // otherwise an error code.
+  virtual int RegisterPayload(int rtp_payload_type,
+                              const SdpAudioFormat& audio_format);
+
+  // Removes the entry for `rtp_payload_type` from the database.
+  // Returns kDecoderNotFound or kOK depending on the outcome of the operation.
+  virtual int Remove(uint8_t rtp_payload_type);
+
+  // Remove all entries.
+  virtual void RemoveAll();
+
+  // Returns a pointer to the DecoderInfo struct for `rtp_payload_type`. If
+  // no decoder is registered with that `rtp_payload_type`, NULL is returned.
+  virtual const DecoderInfo* GetDecoderInfo(uint8_t rtp_payload_type) const;
+
+  // Sets the active decoder to be `rtp_payload_type`. If this call results in a
+  // change of active decoder, `new_decoder` is set to true. The previous active
+  // decoder's AudioDecoder object is deleted.
+  virtual int SetActiveDecoder(uint8_t rtp_payload_type, bool* new_decoder);
+
+  // Returns the current active decoder, or NULL if no active decoder exists.
+  virtual AudioDecoder* GetActiveDecoder() const;
+
+  // Sets the active comfort noise decoder to be `rtp_payload_type`. If this
+  // call results in a change of active comfort noise decoder, the previous
+  // active decoder's AudioDecoder object is deleted.
+  virtual int SetActiveCngDecoder(uint8_t rtp_payload_type);
+
+  // Returns the current active comfort noise decoder, or NULL if no active
+  // comfort noise decoder exists.
+  virtual ComfortNoiseDecoder* GetActiveCngDecoder() const;
+
+  // The following are utility methods: they will look up DecoderInfo through
+  // GetDecoderInfo and call the respective method on that info object, if it
+  // exists.
+
+  // Returns a pointer to the AudioDecoder object associated with
+  // `rtp_payload_type`, or NULL if none is registered. If the AudioDecoder
+  // object does not exist for that decoder, the object is created.
+  AudioDecoder* GetDecoder(uint8_t rtp_payload_type) const;
+
+  // Returns true if `rtp_payload_type` is registered as comfort noise.
+  bool IsComfortNoise(uint8_t rtp_payload_type) const;
+
+  // Returns true if `rtp_payload_type` is registered as DTMF.
+  bool IsDtmf(uint8_t rtp_payload_type) const;
+
+  // Returns true if `rtp_payload_type` is registered as RED.
+  bool IsRed(uint8_t rtp_payload_type) const;
+
+  // Returns kOK if all packets in `packet_list` carry payload types that are
+  // registered in the database. Otherwise, returns kDecoderNotFound.
+  int CheckPayloadTypes(const PacketList& packet_list) const;
+
+ private:
+  typedef std::map<uint8_t, DecoderInfo> DecoderMap;
+
+  DecoderMap decoders_;
+  int active_decoder_type_;
+  int active_cng_decoder_type_;
+  mutable std::unique_ptr<ComfortNoiseDecoder> active_cng_decoder_;
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  const absl::optional<AudioCodecPairId> codec_pair_id_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
new file mode 100644
index 0000000000..445c21924b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
@@ -0,0 +1,227 @@
+/*
+ *  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/neteq/decoder_database.h"
+
+#include <stdlib.h>
+
+#include <string>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder.h"
+#include "test/mock_audio_decoder_factory.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+
+namespace webrtc {
+
+TEST(DecoderDatabase, CreateAndDestroy) {
+  DecoderDatabase db(rtc::make_ref_counted<MockAudioDecoderFactory>(),
+                     absl::nullopt);
+  EXPECT_EQ(0, db.Size());
+  EXPECT_TRUE(db.Empty());
+}
+
+TEST(DecoderDatabase, InsertAndRemove) {
+  auto factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  DecoderDatabase db(factory, absl::nullopt);
+  const uint8_t kPayloadType = 0;
+  const std::string kCodecName = "Robert\'); DROP TABLE Students;";
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadType, SdpAudioFormat(kCodecName, 8000, 1)));
+  EXPECT_EQ(1, db.Size());
+  EXPECT_FALSE(db.Empty());
+  EXPECT_EQ(DecoderDatabase::kOK, db.Remove(kPayloadType));
+  EXPECT_EQ(0, db.Size());
+  EXPECT_TRUE(db.Empty());
+}
+
+TEST(DecoderDatabase, InsertAndRemoveAll) {
+  auto factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  DecoderDatabase db(factory, absl::nullopt);
+  const std::string kCodecName1 = "Robert\'); DROP TABLE Students;";
+  const std::string kCodecName2 = "https://xkcd.com/327/";
+  EXPECT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(0, SdpAudioFormat(kCodecName1, 8000, 1)));
+  EXPECT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(1, SdpAudioFormat(kCodecName2, 8000, 1)));
+  EXPECT_EQ(2, db.Size());
+  EXPECT_FALSE(db.Empty());
+  db.RemoveAll();
+  EXPECT_EQ(0, db.Size());
+  EXPECT_TRUE(db.Empty());
+}
+
+TEST(DecoderDatabase, GetDecoderInfo) {
+  auto factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  auto* decoder = new MockAudioDecoder;
+  EXPECT_CALL(*factory, MakeAudioDecoderMock(_, _, _))
+      .WillOnce(Invoke([decoder](const SdpAudioFormat& format,
+                                 absl::optional<AudioCodecPairId> codec_pair_id,
+                                 std::unique_ptr<AudioDecoder>* dec) {
+        EXPECT_EQ("pcmu", format.name);
+        dec->reset(decoder);
+      }));
+  DecoderDatabase db(factory, absl::nullopt);
+  const uint8_t kPayloadType = 0;
+  const std::string kCodecName = "pcmu";
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadType, SdpAudioFormat(kCodecName, 8000, 1)));
+  const DecoderDatabase::DecoderInfo* info;
+  info = db.GetDecoderInfo(kPayloadType);
+  ASSERT_TRUE(info != NULL);
+  EXPECT_TRUE(info->IsType("pcmu"));
+  EXPECT_EQ(kCodecName, info->get_name());
+  EXPECT_EQ(decoder, db.GetDecoder(kPayloadType));
+  info = db.GetDecoderInfo(kPayloadType + 1);  // Other payload type.
+  EXPECT_TRUE(info == NULL);                   // Should not be found.
+}
+
+TEST(DecoderDatabase, GetDecoder) {
+  DecoderDatabase db(CreateBuiltinAudioDecoderFactory(), absl::nullopt);
+  const uint8_t kPayloadType = 0;
+  EXPECT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(kPayloadType, SdpAudioFormat("l16", 8000, 1)));
+  AudioDecoder* dec = db.GetDecoder(kPayloadType);
+  ASSERT_TRUE(dec != NULL);
+}
+
+TEST(DecoderDatabase, TypeTests) {
+  auto factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  DecoderDatabase db(factory, absl::nullopt);
+  const uint8_t kPayloadTypePcmU = 0;
+  const uint8_t kPayloadTypeCng = 13;
+  const uint8_t kPayloadTypeDtmf = 100;
+  const uint8_t kPayloadTypeRed = 101;
+  const uint8_t kPayloadNotUsed = 102;
+  // Load into database.
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadTypePcmU, SdpAudioFormat("pcmu", 8000, 1)));
+  EXPECT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(kPayloadTypeCng, SdpAudioFormat("cn", 8000, 1)));
+  EXPECT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(kPayloadTypeDtmf,
+                               SdpAudioFormat("telephone-event", 8000, 1)));
+  EXPECT_EQ(
+      DecoderDatabase::kOK,
+      db.RegisterPayload(kPayloadTypeRed, SdpAudioFormat("red", 8000, 1)));
+  EXPECT_EQ(4, db.Size());
+  // Test.
+  EXPECT_FALSE(db.IsComfortNoise(kPayloadNotUsed));
+  EXPECT_FALSE(db.IsDtmf(kPayloadNotUsed));
+  EXPECT_FALSE(db.IsRed(kPayloadNotUsed));
+  EXPECT_FALSE(db.IsComfortNoise(kPayloadTypePcmU));
+  EXPECT_FALSE(db.IsDtmf(kPayloadTypePcmU));
+  EXPECT_FALSE(db.IsRed(kPayloadTypePcmU));
+  EXPECT_TRUE(db.IsComfortNoise(kPayloadTypeCng));
+  EXPECT_TRUE(db.IsDtmf(kPayloadTypeDtmf));
+  EXPECT_TRUE(db.IsRed(kPayloadTypeRed));
+}
+
+TEST(DecoderDatabase, CheckPayloadTypes) {
+  constexpr int kNumPayloads = 10;
+  auto factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  DecoderDatabase db(factory, absl::nullopt);
+  // Load a number of payloads into the database. Payload types are 0, 1, ...,
+  // while the decoder type is the same for all payload types (this does not
+  // matter for the test).
+  for (uint8_t payload_type = 0; payload_type < kNumPayloads; ++payload_type) {
+    EXPECT_EQ(
+        DecoderDatabase::kOK,
+        db.RegisterPayload(payload_type, SdpAudioFormat("pcmu", 8000, 1)));
+  }
+  PacketList packet_list;
+  for (int i = 0; i < kNumPayloads + 1; ++i) {
+    // Create packet with payload type `i`. The last packet will have a payload
+    // type that is not registered in the decoder database.
+    Packet packet;
+    packet.payload_type = i;
+    packet_list.push_back(std::move(packet));
+  }
+
+  // Expect to return false, since the last packet is of an unknown type.
+  EXPECT_EQ(DecoderDatabase::kDecoderNotFound,
+            db.CheckPayloadTypes(packet_list));
+
+  packet_list.pop_back();  // Remove the unknown one.
+
+  EXPECT_EQ(DecoderDatabase::kOK, db.CheckPayloadTypes(packet_list));
+
+  // Delete all packets.
+  PacketList::iterator it = packet_list.begin();
+  while (it != packet_list.end()) {
+    it = packet_list.erase(it);
+  }
+}
+
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#define IF_ISAC(x) x
+#else
+#define IF_ISAC(x) DISABLED_##x
+#endif
+
+// Test the methods for setting and getting active speech and CNG decoders.
+TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) {
+  DecoderDatabase db(CreateBuiltinAudioDecoderFactory(), absl::nullopt);
+  // Load payload types.
+  ASSERT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(0, SdpAudioFormat("pcmu", 8000, 1)));
+  ASSERT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(103, SdpAudioFormat("isac", 16000, 1)));
+  ASSERT_EQ(DecoderDatabase::kOK,
+            db.RegisterPayload(13, SdpAudioFormat("cn", 8000, 1)));
+  // Verify that no decoders are active from the start.
+  EXPECT_EQ(NULL, db.GetActiveDecoder());
+  EXPECT_EQ(NULL, db.GetActiveCngDecoder());
+
+  // Set active speech codec.
+  bool changed;  // Should be true when the active decoder changed.
+  EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveDecoder(0, &changed));
+  EXPECT_TRUE(changed);
+  AudioDecoder* decoder = db.GetActiveDecoder();
+  ASSERT_FALSE(decoder == NULL);  // Should get a decoder here.
+
+  // Set the same again. Expect no change.
+  EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveDecoder(0, &changed));
+  EXPECT_FALSE(changed);
+  decoder = db.GetActiveDecoder();
+  ASSERT_FALSE(decoder == NULL);  // Should get a decoder here.
+
+  // Change active decoder.
+  EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveDecoder(103, &changed));
+  EXPECT_TRUE(changed);
+  decoder = db.GetActiveDecoder();
+  ASSERT_FALSE(decoder == NULL);  // Should get a decoder here.
+
+  // Remove the active decoder, and verify that the active becomes NULL.
+  EXPECT_EQ(DecoderDatabase::kOK, db.Remove(103));
+  EXPECT_EQ(NULL, db.GetActiveDecoder());
+
+  // Set active CNG codec.
+  EXPECT_EQ(DecoderDatabase::kOK, db.SetActiveCngDecoder(13));
+  ComfortNoiseDecoder* cng = db.GetActiveCngDecoder();
+  ASSERT_FALSE(cng == NULL);  // Should get a decoder here.
+
+  // Remove the active CNG decoder, and verify that the active becomes NULL.
+  EXPECT_EQ(DecoderDatabase::kOK, db.Remove(13));
+  EXPECT_EQ(NULL, db.GetActiveCngDecoder());
+
+  // Try to set non-existing codecs as active.
+  EXPECT_EQ(DecoderDatabase::kDecoderNotFound,
+            db.SetActiveDecoder(17, &changed));
+  EXPECT_EQ(DecoderDatabase::kDecoderNotFound, db.SetActiveCngDecoder(17));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.cc b/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.cc
new file mode 100644
index 0000000000..487450fe0f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.cc
@@ -0,0 +1,31 @@
+/*
+ *  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/neteq/default_neteq_factory.h"
+
+#include <utility>
+
+#include "modules/audio_coding/neteq/neteq_impl.h"
+
+namespace webrtc {
+
+DefaultNetEqFactory::DefaultNetEqFactory() = default;
+DefaultNetEqFactory::~DefaultNetEqFactory() = default;
+
+std::unique_ptr<NetEq> DefaultNetEqFactory::CreateNetEq(
+    const NetEq::Config& config,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+    Clock* clock) const {
+  return std::make_unique<NetEqImpl>(
+      config, NetEqImpl::Dependencies(config, clock, decoder_factory,
+                                      controller_factory_));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.h b/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.h
new file mode 100644
index 0000000000..24d2bae419
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/default_neteq_factory.h
@@ -0,0 +1,41 @@
+/*
+ *  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_NETEQ_DEFAULT_NETEQ_FACTORY_H_
+#define MODULES_AUDIO_CODING_NETEQ_DEFAULT_NETEQ_FACTORY_H_
+
+#include <memory>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/neteq/default_neteq_controller_factory.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/scoped_refptr.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class DefaultNetEqFactory : public NetEqFactory {
+ public:
+  DefaultNetEqFactory();
+  ~DefaultNetEqFactory() override;
+  DefaultNetEqFactory(const DefaultNetEqFactory&) = delete;
+  DefaultNetEqFactory& operator=(const DefaultNetEqFactory&) = delete;
+
+  std::unique_ptr<NetEq> CreateNetEq(
+      const NetEq::Config& config,
+      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+      Clock* clock) const override;
+
+ private:
+  const DefaultNetEqControllerFactory controller_factory_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DEFAULT_NETEQ_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc
new file mode 100644
index 0000000000..bf3a0f18a1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.cc
@@ -0,0 +1,207 @@
+/*
+ *  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/neteq/delay_manager.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <algorithm>
+#include <memory>
+#include <numeric>
+#include <string>
+
+#include "modules/include/module_common_types_public.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kMinBaseMinimumDelayMs = 0;
+constexpr int kMaxBaseMinimumDelayMs = 10000;
+constexpr int kStartDelayMs = 80;
+
+std::unique_ptr<ReorderOptimizer> MaybeCreateReorderOptimizer(
+    const DelayManager::Config& config) {
+  if (!config.use_reorder_optimizer) {
+    return nullptr;
+  }
+  return std::make_unique<ReorderOptimizer>(
+      (1 << 15) * config.reorder_forget_factor, config.ms_per_loss_percent,
+      config.start_forget_weight);
+}
+
+}  // namespace
+
+DelayManager::Config::Config() {
+  StructParametersParser::Create(                       //
+      "quantile", &quantile,                            //
+      "forget_factor", &forget_factor,                  //
+      "start_forget_weight", &start_forget_weight,      //
+      "resample_interval_ms", &resample_interval_ms,    //
+      "use_reorder_optimizer", &use_reorder_optimizer,  //
+      "reorder_forget_factor", &reorder_forget_factor,  //
+      "ms_per_loss_percent", &ms_per_loss_percent)
+      ->Parse(webrtc::field_trial::FindFullName(
+          "WebRTC-Audio-NetEqDelayManagerConfig"));
+}
+
+void DelayManager::Config::Log() {
+  RTC_LOG(LS_INFO) << "Delay manager config:"
+                      " quantile="
+                   << quantile << " forget_factor=" << forget_factor
+                   << " start_forget_weight=" << start_forget_weight.value_or(0)
+                   << " resample_interval_ms="
+                   << resample_interval_ms.value_or(0)
+                   << " use_reorder_optimizer=" << use_reorder_optimizer
+                   << " reorder_forget_factor=" << reorder_forget_factor
+                   << " ms_per_loss_percent=" << ms_per_loss_percent;
+}
+
+DelayManager::DelayManager(const Config& config, const TickTimer* tick_timer)
+    : max_packets_in_buffer_(config.max_packets_in_buffer),
+      underrun_optimizer_(tick_timer,
+                          (1 << 30) * config.quantile,
+                          (1 << 15) * config.forget_factor,
+                          config.start_forget_weight,
+                          config.resample_interval_ms),
+      reorder_optimizer_(MaybeCreateReorderOptimizer(config)),
+      base_minimum_delay_ms_(config.base_minimum_delay_ms),
+      effective_minimum_delay_ms_(config.base_minimum_delay_ms),
+      minimum_delay_ms_(0),
+      maximum_delay_ms_(0),
+      target_level_ms_(kStartDelayMs) {
+  RTC_DCHECK_GE(base_minimum_delay_ms_, 0);
+
+  Reset();
+}
+
+DelayManager::~DelayManager() {}
+
+void DelayManager::Update(int arrival_delay_ms, bool reordered) {
+  if (!reorder_optimizer_ || !reordered) {
+    underrun_optimizer_.Update(arrival_delay_ms);
+  }
+  target_level_ms_ =
+      underrun_optimizer_.GetOptimalDelayMs().value_or(kStartDelayMs);
+  if (reorder_optimizer_) {
+    reorder_optimizer_->Update(arrival_delay_ms, reordered, target_level_ms_);
+    target_level_ms_ = std::max(
+        target_level_ms_, reorder_optimizer_->GetOptimalDelayMs().value_or(0));
+  }
+  unlimited_target_level_ms_ = target_level_ms_;
+  target_level_ms_ = std::max(target_level_ms_, effective_minimum_delay_ms_);
+  if (maximum_delay_ms_ > 0) {
+    target_level_ms_ = std::min(target_level_ms_, maximum_delay_ms_);
+  }
+  if (packet_len_ms_ > 0) {
+    // Limit to 75% of maximum buffer size.
+    target_level_ms_ = std::min(
+        target_level_ms_, 3 * max_packets_in_buffer_ * packet_len_ms_ / 4);
+  }
+}
+
+int DelayManager::SetPacketAudioLength(int length_ms) {
+  if (length_ms <= 0) {
+    RTC_LOG_F(LS_ERROR) << "length_ms = " << length_ms;
+    return -1;
+  }
+  packet_len_ms_ = length_ms;
+  return 0;
+}
+
+void DelayManager::Reset() {
+  packet_len_ms_ = 0;
+  underrun_optimizer_.Reset();
+  target_level_ms_ = kStartDelayMs;
+  if (reorder_optimizer_) {
+    reorder_optimizer_->Reset();
+  }
+}
+
+int DelayManager::TargetDelayMs() const {
+  return target_level_ms_;
+}
+
+int DelayManager::UnlimitedTargetLevelMs() const {
+  return unlimited_target_level_ms_;
+}
+
+bool DelayManager::IsValidMinimumDelay(int delay_ms) const {
+  return 0 <= delay_ms && delay_ms <= MinimumDelayUpperBound();
+}
+
+bool DelayManager::IsValidBaseMinimumDelay(int delay_ms) const {
+  return kMinBaseMinimumDelayMs <= delay_ms &&
+         delay_ms <= kMaxBaseMinimumDelayMs;
+}
+
+bool DelayManager::SetMinimumDelay(int delay_ms) {
+  if (!IsValidMinimumDelay(delay_ms)) {
+    return false;
+  }
+
+  minimum_delay_ms_ = delay_ms;
+  UpdateEffectiveMinimumDelay();
+  return true;
+}
+
+bool DelayManager::SetMaximumDelay(int delay_ms) {
+  // If `delay_ms` is zero then it unsets the maximum delay and target level is
+  // unconstrained by maximum delay.
+  if (delay_ms != 0 && delay_ms < minimum_delay_ms_) {
+    // Maximum delay shouldn't be less than minimum delay or less than a packet.
+    return false;
+  }
+
+  maximum_delay_ms_ = delay_ms;
+  UpdateEffectiveMinimumDelay();
+  return true;
+}
+
+bool DelayManager::SetBaseMinimumDelay(int delay_ms) {
+  if (!IsValidBaseMinimumDelay(delay_ms)) {
+    return false;
+  }
+
+  base_minimum_delay_ms_ = delay_ms;
+  UpdateEffectiveMinimumDelay();
+  return true;
+}
+
+int DelayManager::GetBaseMinimumDelay() const {
+  return base_minimum_delay_ms_;
+}
+
+void DelayManager::UpdateEffectiveMinimumDelay() {
+  // Clamp `base_minimum_delay_ms_` into the range which can be effectively
+  // used.
+  const int base_minimum_delay_ms =
+      rtc::SafeClamp(base_minimum_delay_ms_, 0, MinimumDelayUpperBound());
+  effective_minimum_delay_ms_ =
+      std::max(minimum_delay_ms_, base_minimum_delay_ms);
+}
+
+int DelayManager::MinimumDelayUpperBound() const {
+  // Choose the lowest possible bound discarding 0 cases which mean the value
+  // is not set and unconstrained.
+  int q75 = max_packets_in_buffer_ * packet_len_ms_ * 3 / 4;
+  q75 = q75 > 0 ? q75 : kMaxBaseMinimumDelayMs;
+  const int maximum_delay_ms =
+      maximum_delay_ms_ > 0 ? maximum_delay_ms_ : kMaxBaseMinimumDelayMs;
+  return std::min(maximum_delay_ms, q75);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.h b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.h
new file mode 100644
index 0000000000..a333681535
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager.h
@@ -0,0 +1,121 @@
+/*
+ *  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_NETEQ_DELAY_MANAGER_H_
+#define MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
+
+#include <string.h>  // Provide access to size_t.
+
+#include <deque>
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/histogram.h"
+#include "modules/audio_coding/neteq/reorder_optimizer.h"
+#include "modules/audio_coding/neteq/underrun_optimizer.h"
+
+namespace webrtc {
+
+class DelayManager {
+ public:
+  struct Config {
+    Config();
+    void Log();
+
+    // Options that can be configured via field trial.
+    double quantile = 0.95;
+    double forget_factor = 0.983;
+    absl::optional<double> start_forget_weight = 2;
+    absl::optional<int> resample_interval_ms = 500;
+
+    bool use_reorder_optimizer = true;
+    double reorder_forget_factor = 0.9993;
+    int ms_per_loss_percent = 20;
+
+    // Options that are externally populated.
+    int max_packets_in_buffer = 200;
+    int base_minimum_delay_ms = 0;
+  };
+
+  DelayManager(const Config& config, const TickTimer* tick_timer);
+
+  virtual ~DelayManager();
+
+  DelayManager(const DelayManager&) = delete;
+  DelayManager& operator=(const DelayManager&) = delete;
+
+  // Updates the delay manager that a new packet arrived with delay
+  // `arrival_delay_ms`. This updates the statistics and a new target buffer
+  // level is calculated. The `reordered` flag indicates if the packet was
+  // reordered.
+  virtual void Update(int arrival_delay_ms, bool reordered);
+
+  // Resets all state.
+  virtual void Reset();
+
+  // Gets the target buffer level in milliseconds. If a minimum or maximum delay
+  // has been set, the target delay reported here also respects the configured
+  // min/max delay.
+  virtual int TargetDelayMs() const;
+
+  // Reports the target delay that would be used if no minimum/maximum delay
+  // would be set.
+  virtual int UnlimitedTargetLevelMs() const;
+
+  // Notifies the DelayManager of how much audio data is carried in each packet.
+  virtual int SetPacketAudioLength(int length_ms);
+
+  // Accessors and mutators.
+  // Assuming `delay` is in valid range.
+  virtual bool SetMinimumDelay(int delay_ms);
+  virtual bool SetMaximumDelay(int delay_ms);
+  virtual bool SetBaseMinimumDelay(int delay_ms);
+  virtual int GetBaseMinimumDelay() const;
+
+  // These accessors are only intended for testing purposes.
+  int effective_minimum_delay_ms_for_test() const {
+    return effective_minimum_delay_ms_;
+  }
+
+ private:
+  // Provides value which minimum delay can't exceed based on current buffer
+  // size and given `maximum_delay_ms_`. Lower bound is a constant 0.
+  int MinimumDelayUpperBound() const;
+
+  // Updates `effective_minimum_delay_ms_` delay based on current
+  // `minimum_delay_ms_`, `base_minimum_delay_ms_` and `maximum_delay_ms_`
+  // and buffer size.
+  void UpdateEffectiveMinimumDelay();
+
+  // Makes sure that `delay_ms` is less than maximum delay, if any maximum
+  // is set. Also, if possible check `delay_ms` to be less than 75% of
+  // `max_packets_in_buffer_`.
+  bool IsValidMinimumDelay(int delay_ms) const;
+
+  bool IsValidBaseMinimumDelay(int delay_ms) const;
+
+  // TODO(jakobi): set maximum buffer delay instead of number of packets.
+  const int max_packets_in_buffer_;
+  UnderrunOptimizer underrun_optimizer_;
+  std::unique_ptr<ReorderOptimizer> reorder_optimizer_;
+
+  int base_minimum_delay_ms_;
+  int effective_minimum_delay_ms_;  // Used as lower bound for target delay.
+  int minimum_delay_ms_;            // Externally set minimum delay.
+  int maximum_delay_ms_;            // Externally set maximum allowed delay.
+
+  int packet_len_ms_ = 0;
+  int target_level_ms_ = 0;  // Currently preferred buffer level.
+  int unlimited_target_level_ms_ = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
new file mode 100644
index 0000000000..da5f53188c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -0,0 +1,246 @@
+/*
+ *  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.
+ */
+
+// Unit tests for DelayManager class.
+
+#include "modules/audio_coding/neteq/delay_manager.h"
+
+#include <math.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/histogram.h"
+#include "modules/audio_coding/neteq/mock/mock_histogram.h"
+#include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
+#include "rtc_base/checks.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kMaxNumberOfPackets = 200;
+constexpr int kTimeStepMs = 10;
+constexpr int kFrameSizeMs = 20;
+constexpr int kMaxBufferSizeMs = kMaxNumberOfPackets * kFrameSizeMs;
+
+}  // namespace
+
+class DelayManagerTest : public ::testing::Test {
+ protected:
+  DelayManagerTest();
+  virtual void SetUp();
+  void Update(int delay);
+  void IncreaseTime(int inc_ms);
+
+  TickTimer tick_timer_;
+  DelayManager dm_;
+};
+
+DelayManagerTest::DelayManagerTest()
+    : dm_(DelayManager::Config(), &tick_timer_) {}
+
+void DelayManagerTest::SetUp() {
+  dm_.SetPacketAudioLength(kFrameSizeMs);
+}
+
+void DelayManagerTest::Update(int delay) {
+  dm_.Update(delay, false);
+}
+
+void DelayManagerTest::IncreaseTime(int inc_ms) {
+  for (int t = 0; t < inc_ms; t += kTimeStepMs) {
+    tick_timer_.Increment();
+  }
+}
+
+TEST_F(DelayManagerTest, CreateAndDestroy) {
+  // Nothing to do here. The test fixture creates and destroys the DelayManager
+  // object.
+}
+
+TEST_F(DelayManagerTest, UpdateNormal) {
+  for (int i = 0; i < 50; ++i) {
+    Update(0);
+    IncreaseTime(kFrameSizeMs);
+  }
+  EXPECT_EQ(20, dm_.TargetDelayMs());
+}
+
+TEST_F(DelayManagerTest, MaxDelay) {
+  Update(0);
+  const int kMaxDelayMs = 60;
+  EXPECT_GT(dm_.TargetDelayMs(), kMaxDelayMs);
+  EXPECT_TRUE(dm_.SetMaximumDelay(kMaxDelayMs));
+  Update(0);
+  EXPECT_EQ(kMaxDelayMs, dm_.TargetDelayMs());
+}
+
+TEST_F(DelayManagerTest, MinDelay) {
+  Update(0);
+  int kMinDelayMs = 7 * kFrameSizeMs;
+  EXPECT_LT(dm_.TargetDelayMs(), kMinDelayMs);
+  dm_.SetMinimumDelay(kMinDelayMs);
+  IncreaseTime(kFrameSizeMs);
+  Update(0);
+  EXPECT_EQ(kMinDelayMs, dm_.TargetDelayMs());
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayCheckValidRange) {
+  // Base minimum delay should be between [0, 10000] milliseconds.
+  EXPECT_FALSE(dm_.SetBaseMinimumDelay(-1));
+  EXPECT_FALSE(dm_.SetBaseMinimumDelay(10001));
+  EXPECT_EQ(dm_.GetBaseMinimumDelay(), 0);
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(7999));
+  EXPECT_EQ(dm_.GetBaseMinimumDelay(), 7999);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayLowerThanMinimumDelay) {
+  constexpr int kBaseMinimumDelayMs = 100;
+  constexpr int kMinimumDelayMs = 200;
+
+  // Base minimum delay sets lower bound on minimum. That is why when base
+  // minimum delay is lower than minimum delay we use minimum delay.
+  RTC_DCHECK_LT(kBaseMinimumDelayMs, kMinimumDelayMs);
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kMinimumDelayMs);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayGreaterThanMinimumDelay) {
+  constexpr int kBaseMinimumDelayMs = 70;
+  constexpr int kMinimumDelayMs = 30;
+
+  // Base minimum delay sets lower bound on minimum. That is why when base
+  // minimum delay is greater than minimum delay we use base minimum delay.
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMinimumDelayMs);
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kBaseMinimumDelayMs);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayGreaterThanBufferSize) {
+  constexpr int kBaseMinimumDelayMs = kMaxBufferSizeMs + 1;
+  constexpr int kMinimumDelayMs = 12;
+  constexpr int kMaximumDelayMs = 20;
+  constexpr int kMaxBufferSizeMsQ75 = 3 * kMaxBufferSizeMs / 4;
+
+  EXPECT_TRUE(dm_.SetMaximumDelay(kMaximumDelayMs));
+
+  // Base minimum delay is greater than minimum delay, that is why we clamp
+  // it to current the highest possible value which is maximum delay.
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMinimumDelayMs);
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMaxBufferSizeMs);
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMaximumDelayMs);
+  RTC_DCHECK_LT(kMaximumDelayMs, kMaxBufferSizeMsQ75);
+
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+
+  // Unset maximum value.
+  EXPECT_TRUE(dm_.SetMaximumDelay(0));
+
+  // With maximum value unset, the highest possible value now is 75% of
+  // currently possible maximum buffer size.
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kMaxBufferSizeMsQ75);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayGreaterThanMaximumDelay) {
+  constexpr int kMaximumDelayMs = 400;
+  constexpr int kBaseMinimumDelayMs = kMaximumDelayMs + 1;
+  constexpr int kMinimumDelayMs = 20;
+
+  // Base minimum delay is greater than minimum delay, that is why we clamp
+  // it to current the highest possible value which is kMaximumDelayMs.
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMinimumDelayMs);
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMaximumDelayMs);
+  RTC_DCHECK_LT(kMaximumDelayMs, kMaxBufferSizeMs);
+
+  EXPECT_TRUE(dm_.SetMaximumDelay(kMaximumDelayMs));
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kMaximumDelayMs);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelayLowerThanMaxSize) {
+  constexpr int kMaximumDelayMs = 400;
+  constexpr int kBaseMinimumDelayMs = kMaximumDelayMs - 1;
+  constexpr int kMinimumDelayMs = 20;
+
+  // Base minimum delay is greater than minimum delay, and lower than maximum
+  // delays that is why it is used.
+  RTC_DCHECK_GT(kBaseMinimumDelayMs, kMinimumDelayMs);
+  RTC_DCHECK_LT(kBaseMinimumDelayMs, kMaximumDelayMs);
+
+  EXPECT_TRUE(dm_.SetMaximumDelay(kMaximumDelayMs));
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kBaseMinimumDelayMs);
+}
+
+TEST_F(DelayManagerTest, MinimumDelayMemorization) {
+  // Check that when we increase base minimum delay to value higher than
+  // minimum delay then minimum delay is still memorized. This allows to
+  // restore effective minimum delay to memorized minimum delay value when we
+  // decrease base minimum delay.
+  constexpr int kBaseMinimumDelayMsLow = 10;
+  constexpr int kMinimumDelayMs = 20;
+  constexpr int kBaseMinimumDelayMsHigh = 30;
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMsLow));
+  EXPECT_TRUE(dm_.SetMinimumDelay(kMinimumDelayMs));
+  // Minimum delay is used as it is higher than base minimum delay.
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kMinimumDelayMs);
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMsHigh));
+  // Base minimum delay is used as it is now higher than minimum delay.
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kBaseMinimumDelayMsHigh);
+
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMsLow));
+  // Check that minimum delay is memorized and is used again.
+  EXPECT_EQ(dm_.effective_minimum_delay_ms_for_test(), kMinimumDelayMs);
+}
+
+TEST_F(DelayManagerTest, BaseMinimumDelay) {
+  // First packet arrival.
+  Update(0);
+
+  constexpr int kBaseMinimumDelayMs = 7 * kFrameSizeMs;
+  EXPECT_LT(dm_.TargetDelayMs(), kBaseMinimumDelayMs);
+  EXPECT_TRUE(dm_.SetBaseMinimumDelay(kBaseMinimumDelayMs));
+  EXPECT_EQ(dm_.GetBaseMinimumDelay(), kBaseMinimumDelayMs);
+
+  IncreaseTime(kFrameSizeMs);
+  Update(0);
+  EXPECT_EQ(dm_.GetBaseMinimumDelay(), kBaseMinimumDelayMs);
+  EXPECT_EQ(kBaseMinimumDelayMs, dm_.TargetDelayMs());
+}
+
+TEST_F(DelayManagerTest, Failures) {
+  // Wrong packet size.
+  EXPECT_EQ(-1, dm_.SetPacketAudioLength(0));
+  EXPECT_EQ(-1, dm_.SetPacketAudioLength(-1));
+
+  // Minimum delay higher than a maximum delay is not accepted.
+  EXPECT_TRUE(dm_.SetMaximumDelay(20));
+  EXPECT_FALSE(dm_.SetMinimumDelay(40));
+
+  // Maximum delay less than minimum delay is not accepted.
+  EXPECT_TRUE(dm_.SetMaximumDelay(100));
+  EXPECT_TRUE(dm_.SetMinimumDelay(80));
+  EXPECT_FALSE(dm_.SetMaximumDelay(60));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.cc
new file mode 100644
index 0000000000..a979f94214
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.cc
@@ -0,0 +1,373 @@
+/*
+ *  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/neteq/dsp_helper.h"
+
+#include <string.h>  // Access to memset.
+
+#include <algorithm>  // Access to min, max.
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+
+namespace webrtc {
+
+// Table of constants used in method DspHelper::ParabolicFit().
+const int16_t DspHelper::kParabolaCoefficients[17][3] = {
+    {120, 32, 64},   {140, 44, 75},   {150, 50, 80},   {160, 57, 85},
+    {180, 72, 96},   {200, 89, 107},  {210, 98, 112},  {220, 108, 117},
+    {240, 128, 128}, {260, 150, 139}, {270, 162, 144}, {280, 174, 149},
+    {300, 200, 160}, {320, 228, 171}, {330, 242, 176}, {340, 257, 181},
+    {360, 288, 192}};
+
+// Filter coefficients used when downsampling from the indicated sample rates
+// (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12. The corresponding Q0
+// values are provided in the comments before each array.
+
+// Q0 values: {0.3, 0.4, 0.3}.
+const int16_t DspHelper::kDownsample8kHzTbl[3] = {1229, 1638, 1229};
+
+// Q0 values: {0.15, 0.2, 0.3, 0.2, 0.15}.
+const int16_t DspHelper::kDownsample16kHzTbl[5] = {614, 819, 1229, 819, 614};
+
+// Q0 values: {0.1425, 0.1251, 0.1525, 0.1628, 0.1525, 0.1251, 0.1425}.
+const int16_t DspHelper::kDownsample32kHzTbl[7] = {584, 512, 625, 667,
+                                                   625, 512, 584};
+
+// Q0 values: {0.2487, 0.0952, 0.1042, 0.1074, 0.1042, 0.0952, 0.2487}.
+const int16_t DspHelper::kDownsample48kHzTbl[7] = {1019, 390, 427, 440,
+                                                   427,  390, 1019};
+
+int DspHelper::RampSignal(const int16_t* input,
+                          size_t length,
+                          int factor,
+                          int increment,
+                          int16_t* output) {
+  int factor_q20 = (factor << 6) + 32;
+  // TODO(hlundin): Add 32 to factor_q20 when converting back to Q14?
+  for (size_t i = 0; i < length; ++i) {
+    output[i] = (factor * input[i] + 8192) >> 14;
+    factor_q20 += increment;
+    factor_q20 = std::max(factor_q20, 0);  // Never go negative.
+    factor = std::min(factor_q20 >> 6, 16384);
+  }
+  return factor;
+}
+
+int DspHelper::RampSignal(int16_t* signal,
+                          size_t length,
+                          int factor,
+                          int increment) {
+  return RampSignal(signal, length, factor, increment, signal);
+}
+
+int DspHelper::RampSignal(AudioVector* signal,
+                          size_t start_index,
+                          size_t length,
+                          int factor,
+                          int increment) {
+  int factor_q20 = (factor << 6) + 32;
+  // TODO(hlundin): Add 32 to factor_q20 when converting back to Q14?
+  for (size_t i = start_index; i < start_index + length; ++i) {
+    (*signal)[i] = (factor * (*signal)[i] + 8192) >> 14;
+    factor_q20 += increment;
+    factor_q20 = std::max(factor_q20, 0);  // Never go negative.
+    factor = std::min(factor_q20 >> 6, 16384);
+  }
+  return factor;
+}
+
+int DspHelper::RampSignal(AudioMultiVector* signal,
+                          size_t start_index,
+                          size_t length,
+                          int factor,
+                          int increment) {
+  RTC_DCHECK_LE(start_index + length, signal->Size());
+  if (start_index + length > signal->Size()) {
+    // Wrong parameters. Do nothing and return the scale factor unaltered.
+    return factor;
+  }
+  int end_factor = 0;
+  // Loop over the channels, starting at the same `factor` each time.
+  for (size_t channel = 0; channel < signal->Channels(); ++channel) {
+    end_factor =
+        RampSignal(&(*signal)[channel], start_index, length, factor, increment);
+  }
+  return end_factor;
+}
+
+void DspHelper::PeakDetection(int16_t* data,
+                              size_t data_length,
+                              size_t num_peaks,
+                              int fs_mult,
+                              size_t* peak_index,
+                              int16_t* peak_value) {
+  size_t min_index = 0;
+  size_t max_index = 0;
+
+  for (size_t i = 0; i <= num_peaks - 1; i++) {
+    if (num_peaks == 1) {
+      // Single peak.  The parabola fit assumes that an extra point is
+      // available; worst case it gets a zero on the high end of the signal.
+      // TODO(hlundin): This can potentially get much worse. It breaks the
+      // API contract, that the length of `data` is `data_length`.
+      data_length++;
+    }
+
+    peak_index[i] = WebRtcSpl_MaxIndexW16(data, data_length - 1);
+
+    if (i != num_peaks - 1) {
+      min_index = (peak_index[i] > 2) ? (peak_index[i] - 2) : 0;
+      max_index = std::min(data_length - 1, peak_index[i] + 2);
+    }
+
+    if ((peak_index[i] != 0) && (peak_index[i] != (data_length - 2))) {
+      ParabolicFit(&data[peak_index[i] - 1], fs_mult, &peak_index[i],
+                   &peak_value[i]);
+    } else {
+      if (peak_index[i] == data_length - 2) {
+        if (data[peak_index[i]] > data[peak_index[i] + 1]) {
+          ParabolicFit(&data[peak_index[i] - 1], fs_mult, &peak_index[i],
+                       &peak_value[i]);
+        } else if (data[peak_index[i]] <= data[peak_index[i] + 1]) {
+          // Linear approximation.
+          peak_value[i] = (data[peak_index[i]] + data[peak_index[i] + 1]) >> 1;
+          peak_index[i] = (peak_index[i] * 2 + 1) * fs_mult;
+        }
+      } else {
+        peak_value[i] = data[peak_index[i]];
+        peak_index[i] = peak_index[i] * 2 * fs_mult;
+      }
+    }
+
+    if (i != num_peaks - 1) {
+      memset(&data[min_index], 0,
+             sizeof(data[0]) * (max_index - min_index + 1));
+    }
+  }
+}
+
+void DspHelper::ParabolicFit(int16_t* signal_points,
+                             int fs_mult,
+                             size_t* peak_index,
+                             int16_t* peak_value) {
+  uint16_t fit_index[13];
+  if (fs_mult == 1) {
+    fit_index[0] = 0;
+    fit_index[1] = 8;
+    fit_index[2] = 16;
+  } else if (fs_mult == 2) {
+    fit_index[0] = 0;
+    fit_index[1] = 4;
+    fit_index[2] = 8;
+    fit_index[3] = 12;
+    fit_index[4] = 16;
+  } else if (fs_mult == 4) {
+    fit_index[0] = 0;
+    fit_index[1] = 2;
+    fit_index[2] = 4;
+    fit_index[3] = 6;
+    fit_index[4] = 8;
+    fit_index[5] = 10;
+    fit_index[6] = 12;
+    fit_index[7] = 14;
+    fit_index[8] = 16;
+  } else {
+    fit_index[0] = 0;
+    fit_index[1] = 1;
+    fit_index[2] = 3;
+    fit_index[3] = 4;
+    fit_index[4] = 5;
+    fit_index[5] = 7;
+    fit_index[6] = 8;
+    fit_index[7] = 9;
+    fit_index[8] = 11;
+    fit_index[9] = 12;
+    fit_index[10] = 13;
+    fit_index[11] = 15;
+    fit_index[12] = 16;
+  }
+
+  //  num = -3 * signal_points[0] + 4 * signal_points[1] - signal_points[2];
+  //  den =      signal_points[0] - 2 * signal_points[1] + signal_points[2];
+  int32_t num =
+      (signal_points[0] * -3) + (signal_points[1] * 4) - signal_points[2];
+  int32_t den = signal_points[0] + (signal_points[1] * -2) + signal_points[2];
+  int32_t temp = num * 120;
+  int flag = 1;
+  int16_t stp = kParabolaCoefficients[fit_index[fs_mult]][0] -
+                kParabolaCoefficients[fit_index[fs_mult - 1]][0];
+  int16_t strt = (kParabolaCoefficients[fit_index[fs_mult]][0] +
+                  kParabolaCoefficients[fit_index[fs_mult - 1]][0]) /
+                 2;
+  int16_t lmt;
+  if (temp < -den * strt) {
+    lmt = strt - stp;
+    while (flag) {
+      if ((flag == fs_mult) || (temp > -den * lmt)) {
+        *peak_value =
+            (den * kParabolaCoefficients[fit_index[fs_mult - flag]][1] +
+             num * kParabolaCoefficients[fit_index[fs_mult - flag]][2] +
+             signal_points[0] * 256) /
+            256;
+        *peak_index = *peak_index * 2 * fs_mult - flag;
+        flag = 0;
+      } else {
+        flag++;
+        lmt -= stp;
+      }
+    }
+  } else if (temp > -den * (strt + stp)) {
+    lmt = strt + 2 * stp;
+    while (flag) {
+      if ((flag == fs_mult) || (temp < -den * lmt)) {
+        int32_t temp_term_1 =
+            den * kParabolaCoefficients[fit_index[fs_mult + flag]][1];
+        int32_t temp_term_2 =
+            num * kParabolaCoefficients[fit_index[fs_mult + flag]][2];
+        int32_t temp_term_3 = signal_points[0] * 256;
+        *peak_value = (temp_term_1 + temp_term_2 + temp_term_3) / 256;
+        *peak_index = *peak_index * 2 * fs_mult + flag;
+        flag = 0;
+      } else {
+        flag++;
+        lmt += stp;
+      }
+    }
+  } else {
+    *peak_value = signal_points[1];
+    *peak_index = *peak_index * 2 * fs_mult;
+  }
+}
+
+size_t DspHelper::MinDistortion(const int16_t* signal,
+                                size_t min_lag,
+                                size_t max_lag,
+                                size_t length,
+                                int32_t* distortion_value) {
+  size_t best_index = 0;
+  int32_t min_distortion = WEBRTC_SPL_WORD32_MAX;
+  for (size_t i = min_lag; i <= max_lag; i++) {
+    int32_t sum_diff = 0;
+    const int16_t* data1 = signal;
+    const int16_t* data2 = signal - i;
+    for (size_t j = 0; j < length; j++) {
+      sum_diff += WEBRTC_SPL_ABS_W32(data1[j] - data2[j]);
+    }
+    // Compare with previous minimum.
+    if (sum_diff < min_distortion) {
+      min_distortion = sum_diff;
+      best_index = i;
+    }
+  }
+  *distortion_value = min_distortion;
+  return best_index;
+}
+
+void DspHelper::CrossFade(const int16_t* input1,
+                          const int16_t* input2,
+                          size_t length,
+                          int16_t* mix_factor,
+                          int16_t factor_decrement,
+                          int16_t* output) {
+  int16_t factor = *mix_factor;
+  int16_t complement_factor = 16384 - factor;
+  for (size_t i = 0; i < length; i++) {
+    output[i] =
+        (factor * input1[i] + complement_factor * input2[i] + 8192) >> 14;
+    factor -= factor_decrement;
+    complement_factor += factor_decrement;
+  }
+  *mix_factor = factor;
+}
+
+void DspHelper::UnmuteSignal(const int16_t* input,
+                             size_t length,
+                             int16_t* factor,
+                             int increment,
+                             int16_t* output) {
+  uint16_t factor_16b = *factor;
+  int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32;
+  for (size_t i = 0; i < length; i++) {
+    output[i] = (factor_16b * input[i] + 8192) >> 14;
+    factor_32b = std::max(factor_32b + increment, 0);
+    factor_16b = std::min(16384, factor_32b >> 6);
+  }
+  *factor = factor_16b;
+}
+
+void DspHelper::MuteSignal(int16_t* signal, int mute_slope, size_t length) {
+  int32_t factor = (16384 << 6) + 32;
+  for (size_t i = 0; i < length; i++) {
+    signal[i] = ((factor >> 6) * signal[i] + 8192) >> 14;
+    factor -= mute_slope;
+  }
+}
+
+int DspHelper::DownsampleTo4kHz(const int16_t* input,
+                                size_t input_length,
+                                size_t output_length,
+                                int input_rate_hz,
+                                bool compensate_delay,
+                                int16_t* output) {
+  // Set filter parameters depending on input frequency.
+  // NOTE: The phase delay values are wrong compared to the true phase delay
+  // of the filters. However, the error is preserved (through the +1 term) for
+  // consistency.
+  const int16_t* filter_coefficients;  // Filter coefficients.
+  size_t filter_length;                // Number of coefficients.
+  size_t filter_delay;                 // Phase delay in samples.
+  int16_t factor;                      // Conversion rate (inFsHz / 8000).
+  switch (input_rate_hz) {
+    case 8000: {
+      filter_length = 3;
+      factor = 2;
+      filter_coefficients = kDownsample8kHzTbl;
+      filter_delay = 1 + 1;
+      break;
+    }
+    case 16000: {
+      filter_length = 5;
+      factor = 4;
+      filter_coefficients = kDownsample16kHzTbl;
+      filter_delay = 2 + 1;
+      break;
+    }
+    case 32000: {
+      filter_length = 7;
+      factor = 8;
+      filter_coefficients = kDownsample32kHzTbl;
+      filter_delay = 3 + 1;
+      break;
+    }
+    case 48000: {
+      filter_length = 7;
+      factor = 12;
+      filter_coefficients = kDownsample48kHzTbl;
+      filter_delay = 3 + 1;
+      break;
+    }
+    default: {
+      RTC_DCHECK_NOTREACHED();
+      return -1;
+    }
+  }
+
+  if (!compensate_delay) {
+    // Disregard delay compensation.
+    filter_delay = 0;
+  }
+
+  // Returns -1 if input signal is too short; 0 otherwise.
+  return WebRtcSpl_DownsampleFast(
+      &input[filter_length - 1], input_length - filter_length + 1, output,
+      output_length, filter_coefficients, filter_length, factor, filter_delay);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.h b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.h
new file mode 100644
index 0000000000..4aead7df18
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper.h
@@ -0,0 +1,161 @@
+/*
+ *  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_NETEQ_DSP_HELPER_H_
+#define MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
+
+#include <stdint.h>
+#include <string.h>
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
+
+namespace webrtc {
+
+// This class contains various signal processing functions, all implemented as
+// static methods.
+class DspHelper {
+ public:
+  // Filter coefficients used when downsampling from the indicated sample rates
+  // (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12.
+  static const int16_t kDownsample8kHzTbl[3];
+  static const int16_t kDownsample16kHzTbl[5];
+  static const int16_t kDownsample32kHzTbl[7];
+  static const int16_t kDownsample48kHzTbl[7];
+
+  // Constants used to mute and unmute over 5 samples. The coefficients are
+  // in Q15.
+  static const int kMuteFactorStart8kHz = 27307;
+  static const int kMuteFactorIncrement8kHz = -5461;
+  static const int kUnmuteFactorStart8kHz = 5461;
+  static const int kUnmuteFactorIncrement8kHz = 5461;
+  static const int kMuteFactorStart16kHz = 29789;
+  static const int kMuteFactorIncrement16kHz = -2979;
+  static const int kUnmuteFactorStart16kHz = 2979;
+  static const int kUnmuteFactorIncrement16kHz = 2979;
+  static const int kMuteFactorStart32kHz = 31208;
+  static const int kMuteFactorIncrement32kHz = -1560;
+  static const int kUnmuteFactorStart32kHz = 1560;
+  static const int kUnmuteFactorIncrement32kHz = 1560;
+  static const int kMuteFactorStart48kHz = 31711;
+  static const int kMuteFactorIncrement48kHz = -1057;
+  static const int kUnmuteFactorStart48kHz = 1057;
+  static const int kUnmuteFactorIncrement48kHz = 1057;
+
+  // Multiplies the signal with a gradually changing factor.
+  // The first sample is multiplied with `factor` (in Q14). For each sample,
+  // `factor` is increased (additive) by the `increment` (in Q20), which can
+  // be negative. Returns the scale factor after the last increment.
+  static int RampSignal(const int16_t* input,
+                        size_t length,
+                        int factor,
+                        int increment,
+                        int16_t* output);
+
+  // Same as above, but with the samples of `signal` being modified in-place.
+  static int RampSignal(int16_t* signal,
+                        size_t length,
+                        int factor,
+                        int increment);
+
+  // Same as above, but processes `length` samples from `signal`, starting at
+  // `start_index`.
+  static int RampSignal(AudioVector* signal,
+                        size_t start_index,
+                        size_t length,
+                        int factor,
+                        int increment);
+
+  // Same as above, but for an AudioMultiVector.
+  static int RampSignal(AudioMultiVector* signal,
+                        size_t start_index,
+                        size_t length,
+                        int factor,
+                        int increment);
+
+  // Peak detection with parabolic fit. Looks for `num_peaks` maxima in `data`,
+  // having length `data_length` and sample rate multiplier `fs_mult`. The peak
+  // locations and values are written to the arrays `peak_index` and
+  // `peak_value`, respectively. Both arrays must hold at least `num_peaks`
+  // elements.
+  static void PeakDetection(int16_t* data,
+                            size_t data_length,
+                            size_t num_peaks,
+                            int fs_mult,
+                            size_t* peak_index,
+                            int16_t* peak_value);
+
+  // Estimates the height and location of a maximum. The three values in the
+  // array `signal_points` are used as basis for a parabolic fit, which is then
+  // used to find the maximum in an interpolated signal. The `signal_points` are
+  // assumed to be from a 4 kHz signal, while the maximum, written to
+  // `peak_index` and `peak_value` is given in the full sample rate, as
+  // indicated by the sample rate multiplier `fs_mult`.
+  static void ParabolicFit(int16_t* signal_points,
+                           int fs_mult,
+                           size_t* peak_index,
+                           int16_t* peak_value);
+
+  // Calculates the sum-abs-diff for `signal` when compared to a displaced
+  // version of itself. Returns the displacement lag that results in the minimum
+  // distortion. The resulting distortion is written to `distortion_value`.
+  // The values of `min_lag` and `max_lag` are boundaries for the search.
+  static size_t MinDistortion(const int16_t* signal,
+                              size_t min_lag,
+                              size_t max_lag,
+                              size_t length,
+                              int32_t* distortion_value);
+
+  // Mixes `length` samples from `input1` and `input2` together and writes the
+  // result to `output`. The gain for `input1` starts at `mix_factor` (Q14) and
+  // is decreased by `factor_decrement` (Q14) for each sample. The gain for
+  // `input2` is the complement 16384 - mix_factor.
+  static void CrossFade(const int16_t* input1,
+                        const int16_t* input2,
+                        size_t length,
+                        int16_t* mix_factor,
+                        int16_t factor_decrement,
+                        int16_t* output);
+
+  // Scales `input` with an increasing gain. Applies `factor` (Q14) to the first
+  // sample and increases the gain by `increment` (Q20) for each sample. The
+  // result is written to `output`. `length` samples are processed.
+  static void UnmuteSignal(const int16_t* input,
+                           size_t length,
+                           int16_t* factor,
+                           int increment,
+                           int16_t* output);
+
+  // Starts at unity gain and gradually fades out `signal`. For each sample,
+  // the gain is reduced by `mute_slope` (Q14). `length` samples are processed.
+  static void MuteSignal(int16_t* signal, int mute_slope, size_t length);
+
+  // Downsamples `input` from `sample_rate_hz` to 4 kHz sample rate. The input
+  // has `input_length` samples, and the method will write `output_length`
+  // samples to `output`. Compensates for the phase delay of the downsampling
+  // filters if `compensate_delay` is true. Returns -1 if the input is too short
+  // to produce `output_length` samples, otherwise 0.
+  static int DownsampleTo4kHz(const int16_t* input,
+                              size_t input_length,
+                              size_t output_length,
+                              int input_rate_hz,
+                              bool compensate_delay,
+                              int16_t* output);
+
+  DspHelper(const DspHelper&) = delete;
+  DspHelper& operator=(const DspHelper&) = delete;
+
+ private:
+  // Table of constants used in method DspHelper::ParabolicFit().
+  static const int16_t kParabolaCoefficients[17][3];
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc
new file mode 100644
index 0000000000..09247417d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc
@@ -0,0 +1,88 @@
+/*
+ *  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/neteq/dsp_helper.h"
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(DspHelper, RampSignalArray) {
+  static const int kLen = 100;
+  int16_t input[kLen];
+  int16_t output[kLen];
+  // Fill input with 1000.
+  for (int i = 0; i < kLen; ++i) {
+    input[i] = 1000;
+  }
+  int start_factor = 0;
+  // Ramp from 0 to 1 (in Q14) over the array. Note that `increment` is in Q20,
+  // while the factor is in Q14, hence the shift by 6.
+  int increment = (16384 << 6) / kLen;
+
+  // Test first method.
+  int stop_factor =
+      DspHelper::RampSignal(input, kLen, start_factor, increment, output);
+  EXPECT_EQ(16383, stop_factor);  // Almost reach 1 in Q14.
+  for (int i = 0; i < kLen; ++i) {
+    EXPECT_EQ(1000 * i / kLen, output[i]);
+  }
+
+  // Test second method. (Note that this modifies `input`.)
+  stop_factor = DspHelper::RampSignal(input, kLen, start_factor, increment);
+  EXPECT_EQ(16383, stop_factor);  // Almost reach 1 in Q14.
+  for (int i = 0; i < kLen; ++i) {
+    EXPECT_EQ(1000 * i / kLen, input[i]);
+  }
+}
+
+TEST(DspHelper, RampSignalAudioMultiVector) {
+  static const int kLen = 100;
+  static const int kChannels = 5;
+  AudioMultiVector input(kChannels, kLen * 3);
+  // Fill input with 1000.
+  for (int i = 0; i < kLen * 3; ++i) {
+    for (int channel = 0; channel < kChannels; ++channel) {
+      input[channel][i] = 1000;
+    }
+  }
+  // We want to start ramping at `start_index` and keep ramping for `kLen`
+  // samples.
+  int start_index = kLen;
+  int start_factor = 0;
+  // Ramp from 0 to 1 (in Q14) in `kLen` samples. Note that `increment` is in
+  // Q20, while the factor is in Q14, hence the shift by 6.
+  int increment = (16384 << 6) / kLen;
+
+  int stop_factor =
+      DspHelper::RampSignal(&input, start_index, kLen, start_factor, increment);
+  EXPECT_EQ(16383, stop_factor);  // Almost reach 1 in Q14.
+  // Verify that the first `kLen` samples are left untouched.
+  int i;
+  for (i = 0; i < kLen; ++i) {
+    for (int channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(1000, input[channel][i]);
+    }
+  }
+  // Verify that the next block of `kLen` samples are ramped.
+  for (; i < 2 * kLen; ++i) {
+    for (int channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(1000 * (i - kLen) / kLen, input[channel][i]);
+    }
+  }
+  // Verify the last `kLen` samples are left untouched.
+  for (; i < 3 * kLen; ++i) {
+    for (int channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(1000, input[channel][i]);
+    }
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.cc
new file mode 100644
index 0000000000..115bfcf97b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.cc
@@ -0,0 +1,246 @@
+/*
+ *  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/neteq/dtmf_buffer.h"
+
+#include <algorithm>  // max
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+// Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
+// longer required, this #define should be removed (and the code that it
+// enables).
+#define LEGACY_BITEXACT
+
+namespace webrtc {
+
+DtmfBuffer::DtmfBuffer(int fs_hz) {
+  SetSampleRate(fs_hz);
+}
+
+DtmfBuffer::~DtmfBuffer() = default;
+
+void DtmfBuffer::Flush() {
+  buffer_.clear();
+}
+
+// The ParseEvent method parses 4 bytes from `payload` according to this format
+// from RFC 4733:
+//
+//  0                   1                   2                   3
+//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |     event     |E|R| volume    |          duration             |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Legend (adapted from RFC 4733)
+// - event:    The event field is a number between 0 and 255 identifying a
+//             specific telephony event. The buffer will not accept any event
+//             numbers larger than 15.
+// - E:        If set to a value of one, the "end" bit indicates that this
+//             packet contains the end of the event.  For long-lasting events
+//             that have to be split into segments, only the final packet for
+//             the final segment will have the E bit set.
+// - R:        Reserved.
+// - volume:   For DTMF digits and other events representable as tones, this
+//             field describes the power level of the tone, expressed in dBm0
+//             after dropping the sign.  Power levels range from 0 to -63 dBm0.
+//             Thus, larger values denote lower volume. The buffer discards
+//             values larger than 36 (i.e., lower than -36 dBm0).
+// - duration: The duration field indicates the duration of the event or segment
+//             being reported, in timestamp units, expressed as an unsigned
+//             integer in network byte order.  For a non-zero value, the event
+//             or segment began at the instant identified by the RTP timestamp
+//             and has so far lasted as long as indicated by this parameter.
+//             The event may or may not have ended.  If the event duration
+//             exceeds the maximum representable by the duration field, the
+//             event is split into several contiguous segments. The buffer will
+//             discard zero-duration events.
+//
+int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp,
+                           const uint8_t* payload,
+                           size_t payload_length_bytes,
+                           DtmfEvent* event) {
+  RTC_CHECK(payload);
+  RTC_CHECK(event);
+  if (payload_length_bytes < 4) {
+    RTC_LOG(LS_WARNING) << "ParseEvent payload too short";
+    return kPayloadTooShort;
+  }
+
+  event->event_no = payload[0];
+  event->end_bit = ((payload[1] & 0x80) != 0);
+  event->volume = (payload[1] & 0x3F);
+  event->duration = payload[2] << 8 | payload[3];
+  event->timestamp = rtp_timestamp;
+  return kOK;
+}
+
+// Inserts a DTMF event into the buffer. The event should be parsed from the
+// bit stream using the ParseEvent method above before inserting it in the
+// buffer.
+// DTMF events can be quite long, and in most cases the duration of the event
+// is not known when the first packet describing it is sent. To deal with that,
+// the RFC 4733 specifies that multiple packets are sent for one and the same
+// event as it is being created (typically, as the user is pressing the key).
+// These packets will all share the same start timestamp and event number,
+// while the duration will be the cumulative duration from the start. When
+// inserting a new event, the InsertEvent method tries to find a matching event
+// already in the buffer. If so, the new event is simply merged with the
+// existing one.
+int DtmfBuffer::InsertEvent(const DtmfEvent& event) {
+  if (event.event_no < 0 || event.event_no > 15 || event.volume < 0 ||
+      event.volume > 63 || event.duration <= 0 || event.duration > 65535) {
+    RTC_LOG(LS_WARNING) << "InsertEvent invalid parameters";
+    return kInvalidEventParameters;
+  }
+  DtmfList::iterator it = buffer_.begin();
+  while (it != buffer_.end()) {
+    if (MergeEvents(it, event)) {
+      // A matching event was found and the new event was merged.
+      return kOK;
+    }
+    ++it;
+  }
+  buffer_.push_back(event);
+  // Sort the buffer using CompareEvents to rank the events.
+  buffer_.sort(CompareEvents);
+  return kOK;
+}
+
+bool DtmfBuffer::GetEvent(uint32_t current_timestamp, DtmfEvent* event) {
+  DtmfList::iterator it = buffer_.begin();
+  while (it != buffer_.end()) {
+    // `event_end` is an estimate of where the current event ends. If the end
+    // bit is set, we know that the event ends at `timestamp` + `duration`.
+    uint32_t event_end = it->timestamp + it->duration;
+#ifdef LEGACY_BITEXACT
+    bool next_available = false;
+#endif
+    if (!it->end_bit) {
+      // If the end bit is not set, we allow extrapolation of the event for
+      // some time.
+      event_end += max_extrapolation_samples_;
+      DtmfList::iterator next = it;
+      ++next;
+      if (next != buffer_.end()) {
+        // If there is a next event in the buffer, we will not extrapolate over
+        // the start of that new event.
+        event_end = std::min(event_end, next->timestamp);
+#ifdef LEGACY_BITEXACT
+        next_available = true;
+#endif
+      }
+    }
+    if (current_timestamp >= it->timestamp &&
+        current_timestamp <= event_end) {  // TODO(hlundin): Change to <.
+      // Found a matching event.
+      if (event) {
+        event->event_no = it->event_no;
+        event->end_bit = it->end_bit;
+        event->volume = it->volume;
+        event->duration = it->duration;
+        event->timestamp = it->timestamp;
+      }
+#ifdef LEGACY_BITEXACT
+      if (it->end_bit && current_timestamp + frame_len_samples_ >= event_end) {
+        // We are done playing this. Erase the event.
+        buffer_.erase(it);
+      }
+#endif
+      return true;
+    } else if (current_timestamp > event_end) {  // TODO(hlundin): Change to >=.
+// Erase old event. Operation returns a valid pointer to the next element
+// in the list.
+#ifdef LEGACY_BITEXACT
+      if (!next_available) {
+        if (event) {
+          event->event_no = it->event_no;
+          event->end_bit = it->end_bit;
+          event->volume = it->volume;
+          event->duration = it->duration;
+          event->timestamp = it->timestamp;
+        }
+        it = buffer_.erase(it);
+        return true;
+      } else {
+        it = buffer_.erase(it);
+      }
+#else
+      it = buffer_.erase(it);
+#endif
+    } else {
+      ++it;
+    }
+  }
+  return false;
+}
+
+size_t DtmfBuffer::Length() const {
+  return buffer_.size();
+}
+
+bool DtmfBuffer::Empty() const {
+  return buffer_.empty();
+}
+
+int DtmfBuffer::SetSampleRate(int fs_hz) {
+  if (fs_hz != 8000 &&
+      fs_hz != 16000 &&
+      fs_hz != 32000 &&
+      fs_hz != 44100 &&
+      fs_hz != 48000) {
+    return kInvalidSampleRate;
+  }
+  max_extrapolation_samples_ = 7 * fs_hz / 100;
+  frame_len_samples_ = fs_hz / 100;
+  return kOK;
+}
+
+// The method returns true if the two events are considered to be the same.
+// The are defined as equal if they share the same timestamp and event number.
+// The special case with long-lasting events that have to be split into segments
+// is not handled in this method. These will be treated as separate events in
+// the buffer.
+bool DtmfBuffer::SameEvent(const DtmfEvent& a, const DtmfEvent& b) {
+  return (a.event_no == b.event_no) && (a.timestamp == b.timestamp);
+}
+
+bool DtmfBuffer::MergeEvents(DtmfList::iterator it, const DtmfEvent& event) {
+  if (SameEvent(*it, event)) {
+    if (!it->end_bit) {
+      // Do not extend the duration of an event for which the end bit was
+      // already received.
+      it->duration = std::max(event.duration, it->duration);
+    }
+    if (event.end_bit) {
+      it->end_bit = true;
+    }
+    return true;
+  } else {
+    return false;
+  }
+}
+
+// Returns true if `a` goes before `b` in the sorting order ("`a` < `b`").
+// The events are ranked using their start timestamp (taking wrap-around into
+// account). In the unlikely situation that two events share the same start
+// timestamp, the event number is used to rank the two. Note that packets
+// that belong to the same events, and therefore sharing the same start
+// timestamp, have already been merged before the sort method is called.
+bool DtmfBuffer::CompareEvents(const DtmfEvent& a, const DtmfEvent& b) {
+  if (a.timestamp == b.timestamp) {
+    return a.event_no < b.event_no;
+  }
+  // Take wrap-around into account.
+  return (static_cast<uint32_t>(b.timestamp - a.timestamp) < 0xFFFFFFFF / 2);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.h
new file mode 100644
index 0000000000..62b751525c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer.h
@@ -0,0 +1,104 @@
+/*
+ *  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_NETEQ_DTMF_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+
+namespace webrtc {
+
+struct DtmfEvent {
+  uint32_t timestamp;
+  int event_no;
+  int volume;
+  int duration;
+  bool end_bit;
+
+  // Constructors
+  DtmfEvent()
+      : timestamp(0), event_no(0), volume(0), duration(0), end_bit(false) {}
+  DtmfEvent(uint32_t ts, int ev, int vol, int dur, bool end)
+      : timestamp(ts), event_no(ev), volume(vol), duration(dur), end_bit(end) {}
+};
+
+// This is the buffer holding DTMF events while waiting for them to be played.
+class DtmfBuffer {
+ public:
+  enum BufferReturnCodes {
+    kOK = 0,
+    kInvalidPointer,
+    kPayloadTooShort,
+    kInvalidEventParameters,
+    kInvalidSampleRate
+  };
+
+  // Set up the buffer for use at sample rate `fs_hz`.
+  explicit DtmfBuffer(int fs_hz);
+
+  virtual ~DtmfBuffer();
+
+  DtmfBuffer(const DtmfBuffer&) = delete;
+  DtmfBuffer& operator=(const DtmfBuffer&) = delete;
+
+  // Flushes the buffer.
+  virtual void Flush();
+
+  // Static method to parse 4 bytes from `payload` as a DTMF event (RFC 4733)
+  // and write the parsed information into the struct `event`. Input variable
+  // `rtp_timestamp` is simply copied into the struct.
+  static int ParseEvent(uint32_t rtp_timestamp,
+                        const uint8_t* payload,
+                        size_t payload_length_bytes,
+                        DtmfEvent* event);
+
+  // Inserts `event` into the buffer. The method looks for a matching event and
+  // merges the two if a match is found.
+  virtual int InsertEvent(const DtmfEvent& event);
+
+  // Checks if a DTMF event should be played at time `current_timestamp`. If so,
+  // the method returns true; otherwise false. The parameters of the event to
+  // play will be written to `event`.
+  virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event);
+
+  // Number of events in the buffer.
+  virtual size_t Length() const;
+
+  virtual bool Empty() const;
+
+  // Set a new sample rate.
+  virtual int SetSampleRate(int fs_hz);
+
+ private:
+  typedef std::list<DtmfEvent> DtmfList;
+
+  int max_extrapolation_samples_;
+  int frame_len_samples_;  // TODO(hlundin): Remove this later.
+
+  // Compares two events and returns true if they are the same.
+  static bool SameEvent(const DtmfEvent& a, const DtmfEvent& b);
+
+  // Merges `event` to the event pointed out by `it`. The method checks that
+  // the two events are the same (using the SameEvent method), and merges them
+  // if that was the case, returning true. If the events are not the same, false
+  // is returned.
+  bool MergeEvents(DtmfList::iterator it, const DtmfEvent& event);
+
+  // Method used by the sort algorithm to rank events in the buffer.
+  static bool CompareEvents(const DtmfEvent& a, const DtmfEvent& b);
+
+  DtmfList buffer_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
new file mode 100644
index 0000000000..83745b6c09
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
@@ -0,0 +1,297 @@
+/*
+ *  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/neteq/dtmf_buffer.h"
+
+#ifdef WIN32
+#include <winsock2.h>  // ntohl()
+#else
+#include <arpa/inet.h>  // ntohl()
+#endif
+
+#include <iostream>
+
+#include "test/gtest.h"
+
+// Modify the tests so that they pass with the modifications done to DtmfBuffer
+// for backwards bit-exactness. Once bit-exactness is no longer required, this
+// #define should be removed (and the code that it enables).
+#define LEGACY_BITEXACT
+
+namespace webrtc {
+
+static int sample_rate_hz = 8000;
+
+static uint32_t MakeDtmfPayload(int event, bool end, int volume, int duration) {
+  uint32_t payload = 0;
+  //  0                   1                   2                   3
+  //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |     event     |E|R| volume    |          duration             |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  payload |= (event & 0x00FF) << 24;
+  payload |= (end ? 0x00800000 : 0x00000000);
+  payload |= (volume & 0x003F) << 16;
+  payload |= (duration & 0xFFFF);
+  payload = ntohl(payload);
+  return payload;
+}
+
+static bool EqualEvents(const DtmfEvent& a, const DtmfEvent& b) {
+  return (a.duration == b.duration && a.end_bit == b.end_bit &&
+          a.event_no == b.event_no && a.timestamp == b.timestamp &&
+          a.volume == b.volume);
+}
+
+TEST(DtmfBuffer, CreateAndDestroy) {
+  DtmfBuffer* buffer = new DtmfBuffer(sample_rate_hz);
+  delete buffer;
+}
+
+// Test the event parser.
+TEST(DtmfBuffer, ParseEvent) {
+  int event_no = 7;
+  bool end_bit = true;
+  int volume = 17;
+  int duration = 4711;
+  uint32_t timestamp = 0x12345678;
+  uint32_t payload = MakeDtmfPayload(event_no, end_bit, volume, duration);
+  uint8_t* payload_ptr = reinterpret_cast<uint8_t*>(&payload);
+  DtmfEvent event;
+  EXPECT_EQ(DtmfBuffer::kOK, DtmfBuffer::ParseEvent(timestamp, payload_ptr,
+                                                    sizeof(payload), &event));
+  EXPECT_EQ(duration, event.duration);
+  EXPECT_EQ(end_bit, event.end_bit);
+  EXPECT_EQ(event_no, event.event_no);
+  EXPECT_EQ(timestamp, event.timestamp);
+  EXPECT_EQ(volume, event.volume);
+
+  EXPECT_EQ(DtmfBuffer::kPayloadTooShort,
+            DtmfBuffer::ParseEvent(timestamp, payload_ptr, 3, &event));
+}
+
+TEST(DtmfBuffer, SimpleInsertAndGet) {
+  int event_no = 7;
+  bool end_bit = true;
+  int volume = 17;
+  int duration = 4711;
+  uint32_t timestamp = 0x12345678;
+  DtmfEvent event(timestamp, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
+  EXPECT_EQ(1u, buffer.Length());
+  EXPECT_FALSE(buffer.Empty());
+  DtmfEvent out_event;
+  // Too early to get event.
+  EXPECT_FALSE(buffer.GetEvent(timestamp - 10, &out_event));
+  EXPECT_EQ(1u, buffer.Length());
+  EXPECT_FALSE(buffer.Empty());
+  // Get the event at its starting timestamp.
+  EXPECT_TRUE(buffer.GetEvent(timestamp, &out_event));
+  EXPECT_TRUE(EqualEvents(event, out_event));
+  EXPECT_EQ(1u, buffer.Length());
+  EXPECT_FALSE(buffer.Empty());
+  // Get the event some time into the event.
+  EXPECT_TRUE(buffer.GetEvent(timestamp + duration / 2, &out_event));
+  EXPECT_TRUE(EqualEvents(event, out_event));
+  EXPECT_EQ(1u, buffer.Length());
+  EXPECT_FALSE(buffer.Empty());
+// Give a "current" timestamp after the event has ended.
+#ifdef LEGACY_BITEXACT
+  EXPECT_TRUE(buffer.GetEvent(timestamp + duration + 10, &out_event));
+#endif
+  EXPECT_FALSE(buffer.GetEvent(timestamp + duration + 10, &out_event));
+  EXPECT_EQ(0u, buffer.Length());
+  EXPECT_TRUE(buffer.Empty());
+}
+
+TEST(DtmfBuffer, MergingPackets) {
+  int event_no = 0;
+  bool end_bit = false;
+  int volume = 17;
+  int duration = 80;
+  uint32_t timestamp = 0x12345678;
+  DtmfEvent event(timestamp, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
+
+  event.duration += 80;
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
+
+  event.duration += 80;
+  event.end_bit = true;
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
+
+  EXPECT_EQ(1u, buffer.Length());
+
+  DtmfEvent out_event;
+  EXPECT_TRUE(buffer.GetEvent(timestamp, &out_event));
+  EXPECT_TRUE(EqualEvents(event, out_event));
+}
+
+// This test case inserts one shorter event completely overlapped by one longer
+// event. The expected outcome is that only the longer event is played.
+TEST(DtmfBuffer, OverlappingEvents) {
+  int event_no = 0;
+  bool end_bit = true;
+  int volume = 1;
+  int duration = 80;
+  uint32_t timestamp = 0x12345678 + 80;
+  DtmfEvent short_event(timestamp, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(short_event));
+
+  event_no = 10;
+  end_bit = false;
+  timestamp = 0x12345678;
+  DtmfEvent long_event(timestamp, event_no, volume, duration, end_bit);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(long_event));
+
+  long_event.duration += 80;
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(long_event));
+
+  long_event.duration += 80;
+  long_event.end_bit = true;
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(long_event));
+
+  EXPECT_EQ(2u, buffer.Length());
+
+  DtmfEvent out_event;
+  // Expect to get the long event.
+  EXPECT_TRUE(buffer.GetEvent(timestamp, &out_event));
+  EXPECT_TRUE(EqualEvents(long_event, out_event));
+// Expect no more events.
+#ifdef LEGACY_BITEXACT
+  EXPECT_TRUE(
+      buffer.GetEvent(timestamp + long_event.duration + 10, &out_event));
+  EXPECT_TRUE(EqualEvents(long_event, out_event));
+  EXPECT_TRUE(
+      buffer.GetEvent(timestamp + long_event.duration + 10, &out_event));
+  EXPECT_TRUE(EqualEvents(short_event, out_event));
+#else
+  EXPECT_FALSE(
+      buffer.GetEvent(timestamp + long_event.duration + 10, &out_event));
+#endif
+  EXPECT_TRUE(buffer.Empty());
+}
+
+TEST(DtmfBuffer, ExtrapolationTime) {
+  int event_no = 0;
+  bool end_bit = false;
+  int volume = 1;
+  int duration = 80;
+  uint32_t timestamp = 0x12345678;
+  DtmfEvent event1(timestamp, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event1));
+  EXPECT_EQ(1u, buffer.Length());
+
+  DtmfEvent out_event;
+  // Get the event at the start.
+  EXPECT_TRUE(buffer.GetEvent(timestamp, &out_event));
+  EXPECT_TRUE(EqualEvents(event1, out_event));
+  // Also get the event 100 samples after the end of the event (since we're
+  // missing the end bit).
+  uint32_t timestamp_now = timestamp + duration + 100;
+  EXPECT_TRUE(buffer.GetEvent(timestamp_now, &out_event));
+  EXPECT_TRUE(EqualEvents(event1, out_event));
+  // Insert another event starting back-to-back with the previous event.
+  timestamp += duration;
+  event_no = 1;
+  DtmfEvent event2(timestamp, event_no, volume, duration, end_bit);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event2));
+  EXPECT_EQ(2u, buffer.Length());
+  // Now we expect to get the new event when supplying `timestamp_now`.
+  EXPECT_TRUE(buffer.GetEvent(timestamp_now, &out_event));
+  EXPECT_TRUE(EqualEvents(event2, out_event));
+  // Expect the the first event to be erased now.
+  EXPECT_EQ(1u, buffer.Length());
+  // Move `timestamp_now` to more than 560 samples after the end of the second
+  // event. Expect that event to be erased.
+  timestamp_now = timestamp + duration + 600;
+#ifdef LEGACY_BITEXACT
+  EXPECT_TRUE(buffer.GetEvent(timestamp_now, &out_event));
+#endif
+  EXPECT_FALSE(buffer.GetEvent(timestamp_now, &out_event));
+  EXPECT_TRUE(buffer.Empty());
+}
+
+TEST(DtmfBuffer, TimestampWraparound) {
+  int event_no = 0;
+  bool end_bit = true;
+  int volume = 1;
+  int duration = 80;
+  uint32_t timestamp1 = 0xFFFFFFFF - duration;
+  DtmfEvent event1(timestamp1, event_no, volume, duration, end_bit);
+  uint32_t timestamp2 = 0;
+  DtmfEvent event2(timestamp2, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event1));
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event2));
+  EXPECT_EQ(2u, buffer.Length());
+  DtmfEvent out_event;
+  EXPECT_TRUE(buffer.GetEvent(timestamp1, &out_event));
+  EXPECT_TRUE(EqualEvents(event1, out_event));
+#ifdef LEGACY_BITEXACT
+  EXPECT_EQ(1u, buffer.Length());
+#else
+  EXPECT_EQ(2u, buffer.Length());
+#endif
+
+  buffer.Flush();
+  // Reverse the insert order. Expect same results.
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event2));
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event1));
+  EXPECT_EQ(2u, buffer.Length());
+  EXPECT_TRUE(buffer.GetEvent(timestamp1, &out_event));
+  EXPECT_TRUE(EqualEvents(event1, out_event));
+#ifdef LEGACY_BITEXACT
+  EXPECT_EQ(1u, buffer.Length());
+#else
+  EXPECT_EQ(2u, buffer.Length());
+#endif
+}
+
+TEST(DtmfBuffer, InvalidEvents) {
+  int event_no = 0;
+  bool end_bit = true;
+  int volume = 1;
+  int duration = 80;
+  uint32_t timestamp = 0x12345678;
+  DtmfEvent event(timestamp, event_no, volume, duration, end_bit);
+  DtmfBuffer buffer(sample_rate_hz);
+
+  // Invalid event number.
+  event.event_no = -1;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.event_no = 16;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.event_no = 0;  // Valid value;
+
+  // Invalid volume.
+  event.volume = -1;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.volume = 64;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.volume = 0;  // Valid value;
+
+  // Invalid duration.
+  event.duration = -1;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.duration = 0;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.duration = 0xFFFF + 1;
+  EXPECT_EQ(DtmfBuffer::kInvalidEventParameters, buffer.InsertEvent(event));
+  event.duration = 1;  // Valid value;
+
+  // Finish with a valid event, just to verify that all is ok.
+  EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
new file mode 100644
index 0000000000..9061e27c67
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
@@ -0,0 +1,215 @@
+/*
+ *  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 class provides a generator for DTMF tones. The tone generation is based
+// on a sinusoid recursion. Each sinusoid is generated using a recursion
+// formula; x[n] = a * x[n-1] - x[n-2], where the coefficient
+// a = 2*cos(2*pi*f/fs). The recursion is started with x[-1] = 0 and
+// x[-2] = sin(2*pi*f/fs). (Note that with this initialization, the resulting
+// sinusoid gets a "negative" rotation; x[n] = sin(-2*pi*f/fs * n + phi), but
+// kept this way due to historical reasons.)
+// TODO(hlundin): Change to positive rotation?
+//
+// Each key on the telephone keypad corresponds to an "event", 0-15. Each event
+// is mapped to a tone pair, with a low and a high frequency. There are four
+// low and four high frequencies, each corresponding to a row and column,
+// respectively, on the keypad as illustrated below.
+//
+//          1209 Hz  1336 Hz  1477 Hz  1633 Hz
+// 697 Hz      1        2        3       12
+// 770 Hz      4        5        6       13
+// 852 Hz      7        8        9       14
+// 941 Hz     10        0       11       15
+
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
+
+#include "modules/audio_coding/neteq/audio_vector.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// The filter coefficient a = 2*cos(2*pi*f/fs) for the low frequency tone, for
+// sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0 through 15.
+// Values are in Q14.
+const int DtmfToneGenerator::kCoeff1[4][16] = {
+    {24219, 27980, 27980, 27980, 26956, 26956, 26956, 25701, 25701, 25701,
+     24219, 24219, 27980, 26956, 25701, 24219},
+    {30556, 31548, 31548, 31548, 31281, 31281, 31281, 30951, 30951, 30951,
+     30556, 30556, 31548, 31281, 30951, 30556},
+    {32210, 32462, 32462, 32462, 32394, 32394, 32394, 32311, 32311, 32311,
+     32210, 32210, 32462, 32394, 32311, 32210},
+    {32520, 32632, 32632, 32632, 32602, 32602, 32602, 32564, 32564, 32564,
+     32520, 32520, 32632, 32602, 32564, 32520}};
+
+// The filter coefficient a = 2*cos(2*pi*f/fs) for the high frequency tone, for
+// sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0 through 15.
+// Values are in Q14.
+const int DtmfToneGenerator::kCoeff2[4][16] = {
+    {16325, 19073, 16325, 13085, 19073, 16325, 13085, 19073, 16325, 13085,
+     19073, 13085, 9315, 9315, 9315, 9315},
+    {28361, 29144, 28361, 27409, 29144, 28361, 27409, 29144, 28361, 27409,
+     29144, 27409, 26258, 26258, 26258, 26258},
+    {31647, 31849, 31647, 31400, 31849, 31647, 31400, 31849, 31647, 31400,
+     31849, 31400, 31098, 31098, 31098, 31098},
+    {32268, 32359, 32268, 32157, 32359, 32268, 32157, 32359, 32268, 32157,
+     32359, 32157, 32022, 32022, 32022, 32022}};
+
+// The initialization value x[-2] = sin(2*pi*f/fs) for the low frequency tone,
+// for sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0-15.
+// Values are in Q14.
+const int DtmfToneGenerator::kInitValue1[4][16] = {
+    {11036, 8528, 8528, 8528, 9315, 9315, 9315, 10163, 10163, 10163, 11036,
+     11036, 8528, 9315, 10163, 11036},
+    {5918, 4429, 4429, 4429, 4879, 4879, 4879, 5380, 5380, 5380, 5918, 5918,
+     4429, 4879, 5380, 5918},
+    {3010, 2235, 2235, 2235, 2468, 2468, 2468, 2728, 2728, 2728, 3010, 3010,
+     2235, 2468, 2728, 3010},
+    {2013, 1493, 1493, 1493, 1649, 1649, 1649, 1823, 1823, 1823, 2013, 2013,
+     1493, 1649, 1823, 2013}};
+
+// The initialization value x[-2] = sin(2*pi*f/fs) for the high frequency tone,
+// for sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0-15.
+// Values are in Q14.
+const int DtmfToneGenerator::kInitValue2[4][16] = {
+    {14206, 13323, 14206, 15021, 13323, 14206, 15021, 13323, 14206, 15021,
+     13323, 15021, 15708, 15708, 15708, 15708},
+    {8207, 7490, 8207, 8979, 7490, 8207, 8979, 7490, 8207, 8979, 7490, 8979,
+     9801, 9801, 9801, 9801},
+    {4249, 3853, 4249, 4685, 3853, 4249, 4685, 3853, 4249, 4685, 3853, 4685,
+     5164, 5164, 5164, 5164},
+    {2851, 2582, 2851, 3148, 2582, 2851, 3148, 2582, 2851, 3148, 2582, 3148,
+     3476, 3476, 3476, 3476}};
+
+// Amplitude multipliers for volume values 0 through 63, corresponding to
+// 0 dBm0 through -63 dBm0. Values are in Q14.
+// for a in range(0, 64):
+//   print round(16141.0 * 10**(-float(a)/20))
+const int DtmfToneGenerator::kAmplitude[64] = {
+    16141, 14386, 12821, 11427, 10184, 9077, 8090, 7210, 6426, 5727, 5104,
+    4549,  4054,  3614,  3221,  2870,  2558, 2280, 2032, 1811, 1614, 1439,
+    1282,  1143,  1018,  908,   809,   721,  643,  573,  510,  455,  405,
+    361,   322,   287,   256,   228,   203,  181,  161,  144,  128,  114,
+    102,   91,    81,    72,    64,    57,   51,   45,   41,   36,   32,
+    29,    26,    23,    20,    18,    16,   14,   13,   11};
+
+// Constructor.
+DtmfToneGenerator::DtmfToneGenerator()
+    : initialized_(false), coeff1_(0), coeff2_(0), amplitude_(0) {}
+
+// Initialize the DTMF generator with sample rate fs Hz (8000, 16000, 32000,
+// 48000), event (0-15) and attenuation (0-36 dB).
+// Returns 0 on success, otherwise an error code.
+int DtmfToneGenerator::Init(int fs, int event, int attenuation) {
+  initialized_ = false;
+  size_t fs_index;
+  if (fs == 8000) {
+    fs_index = 0;
+  } else if (fs == 16000) {
+    fs_index = 1;
+  } else if (fs == 32000) {
+    fs_index = 2;
+  } else if (fs == 48000) {
+    fs_index = 3;
+  } else {
+    RTC_DCHECK_NOTREACHED();
+    fs_index = 1;  // Default to 8000 Hz.
+  }
+
+  if (event < 0 || event > 15) {
+    return kParameterError;  // Invalid event number.
+  }
+
+  if (attenuation < 0 || attenuation > 63) {
+    return kParameterError;  // Invalid attenuation.
+  }
+
+  // Look up oscillator coefficient for low and high frequencies.
+  RTC_DCHECK_LE(0, fs_index);
+  RTC_DCHECK_GT(arraysize(kCoeff1), fs_index);
+  RTC_DCHECK_GT(arraysize(kCoeff2), fs_index);
+  RTC_DCHECK_LE(0, event);
+  RTC_DCHECK_GT(arraysize(kCoeff1[fs_index]), event);
+  RTC_DCHECK_GT(arraysize(kCoeff2[fs_index]), event);
+  coeff1_ = kCoeff1[fs_index][event];
+  coeff2_ = kCoeff2[fs_index][event];
+
+  // Look up amplitude multiplier.
+  RTC_DCHECK_LE(0, attenuation);
+  RTC_DCHECK_GT(arraysize(kAmplitude), attenuation);
+  amplitude_ = kAmplitude[attenuation];
+
+  // Initialize sample history.
+  RTC_DCHECK_LE(0, fs_index);
+  RTC_DCHECK_GT(arraysize(kInitValue1), fs_index);
+  RTC_DCHECK_GT(arraysize(kInitValue2), fs_index);
+  RTC_DCHECK_LE(0, event);
+  RTC_DCHECK_GT(arraysize(kInitValue1[fs_index]), event);
+  RTC_DCHECK_GT(arraysize(kInitValue2[fs_index]), event);
+  sample_history1_[0] = kInitValue1[fs_index][event];
+  sample_history1_[1] = 0;
+  sample_history2_[0] = kInitValue2[fs_index][event];
+  sample_history2_[1] = 0;
+
+  initialized_ = true;
+  return 0;
+}
+
+// Reset tone generator to uninitialized state.
+void DtmfToneGenerator::Reset() {
+  initialized_ = false;
+}
+
+// Generate num_samples of DTMF signal and write to `output`.
+int DtmfToneGenerator::Generate(size_t num_samples, AudioMultiVector* output) {
+  if (!initialized_) {
+    return kNotInitialized;
+  }
+
+  if (!output) {
+    return kParameterError;
+  }
+
+  output->AssertSize(num_samples);
+  for (size_t i = 0; i < num_samples; ++i) {
+    // Use recursion formula y[n] = a * y[n - 1] - y[n - 2].
+    int16_t temp_val_low =
+        ((coeff1_ * sample_history1_[1] + 8192) >> 14) - sample_history1_[0];
+    int16_t temp_val_high =
+        ((coeff2_ * sample_history2_[1] + 8192) >> 14) - sample_history2_[0];
+
+    // Update recursion memory.
+    sample_history1_[0] = sample_history1_[1];
+    sample_history1_[1] = temp_val_low;
+    sample_history2_[0] = sample_history2_[1];
+    sample_history2_[1] = temp_val_high;
+
+    // Attenuate the low frequency tone 3 dB.
+    int32_t temp_val =
+        kAmpMultiplier * temp_val_low + temp_val_high * (1 << 15);
+    // Normalize the signal to Q14 with proper rounding.
+    temp_val = (temp_val + 16384) >> 15;
+    // Scale the signal to correct volume.
+    (*output)[0][i] =
+        static_cast<int16_t>((temp_val * amplitude_ + 8192) >> 14);
+  }
+  // Copy first channel to all other channels.
+  for (size_t channel = 1; channel < output->Channels(); ++channel) {
+    output->CopyChannel(0, channel);
+  }
+
+  return static_cast<int>(num_samples);
+}
+
+bool DtmfToneGenerator::initialized() const {
+  return initialized_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.h b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
new file mode 100644
index 0000000000..35114f4f49
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
@@ -0,0 +1,57 @@
+/*
+ *  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_NETEQ_DTMF_TONE_GENERATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+namespace webrtc {
+
+// This class provides a generator for DTMF tones.
+class DtmfToneGenerator {
+ public:
+  enum ReturnCodes {
+    kNotInitialized = -1,
+    kParameterError = -2,
+  };
+
+  DtmfToneGenerator();
+  virtual ~DtmfToneGenerator() {}
+
+  DtmfToneGenerator(const DtmfToneGenerator&) = delete;
+  DtmfToneGenerator& operator=(const DtmfToneGenerator&) = delete;
+
+  virtual int Init(int fs, int event, int attenuation);
+  virtual void Reset();
+  virtual int Generate(size_t num_samples, AudioMultiVector* output);
+  virtual bool initialized() const;
+
+ private:
+  static const int kCoeff1[4][16];  // 1st oscillator model coefficient table.
+  static const int kCoeff2[4][16];  // 2nd oscillator model coefficient table.
+  static const int kInitValue1[4][16];  // Initialization for 1st oscillator.
+  static const int kInitValue2[4][16];  // Initialization for 2nd oscillator.
+  static const int kAmplitude[64];      // Amplitude for 0 through -63 dBm0.
+  static const int16_t kAmpMultiplier = 23171;  // 3 dB attenuation (in Q15).
+
+  bool initialized_;            // True if generator is initialized properly.
+  int coeff1_;                  // 1st oscillator coefficient for this event.
+  int coeff2_;                  // 2nd oscillator coefficient for this event.
+  int amplitude_;               // Amplitude for this event.
+  int16_t sample_history1_[2];  // Last 2 samples for the 1st oscillator.
+  int16_t sample_history2_[2];  // Last 2 samples for the 2nd oscillator.
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
new file mode 100644
index 0000000000..e843706dd3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
@@ -0,0 +1,180 @@
+/*
+ *  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.
+ */
+
+// Unit tests for DtmfToneGenerator class.
+
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
+
+#include <math.h>
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class DtmfToneGeneratorTest : public ::testing::Test {
+ protected:
+  static const double kLowFreqHz[16];
+  static const double kHighFreqHz[16];
+  // This is the attenuation applied to all cases.
+  const double kBaseAttenuation = 16141.0 / 16384.0;
+  const double k3dbAttenuation = 23171.0 / 32768;
+  const int kNumSamples = 10;
+
+  void TestAllTones(int fs_hz, int channels) {
+    AudioMultiVector signal(channels);
+
+    for (int event = 0; event <= 15; ++event) {
+      rtc::StringBuilder ss;
+      ss << "Checking event " << event << " at sample rate " << fs_hz;
+      SCOPED_TRACE(ss.str());
+      const int kAttenuation = 0;
+      ASSERT_EQ(0, tone_gen_.Init(fs_hz, event, kAttenuation));
+      EXPECT_TRUE(tone_gen_.initialized());
+      EXPECT_EQ(kNumSamples, tone_gen_.Generate(kNumSamples, &signal));
+
+      double f1 = kLowFreqHz[event];
+      double f2 = kHighFreqHz[event];
+      const double pi = 3.14159265358979323846;
+
+      for (int n = 0; n < kNumSamples; ++n) {
+        double x = k3dbAttenuation * sin(2.0 * pi * f1 / fs_hz * (-n - 1)) +
+                   sin(2.0 * pi * f2 / fs_hz * (-n - 1));
+        x *= kBaseAttenuation;
+        x = ldexp(x, 14);  // Scale to Q14.
+        for (int channel = 0; channel < channels; ++channel) {
+          EXPECT_NEAR(x, static_cast<double>(signal[channel][n]), 25);
+        }
+      }
+
+      tone_gen_.Reset();
+      EXPECT_FALSE(tone_gen_.initialized());
+    }
+  }
+
+  void TestAmplitudes(int fs_hz, int channels) {
+    AudioMultiVector signal(channels);
+    AudioMultiVector ref_signal(channels);
+
+    const int event_vec[] = {0, 4, 9, 13};  // Test a few events.
+    for (int e = 0; e < 4; ++e) {
+      int event = event_vec[e];
+      // Create full-scale reference.
+      ASSERT_EQ(0, tone_gen_.Init(fs_hz, event, 0));  // 0 attenuation.
+      EXPECT_EQ(kNumSamples, tone_gen_.Generate(kNumSamples, &ref_signal));
+      // Test every 5 steps (to save time).
+      for (int attenuation = 1; attenuation <= 63; attenuation += 5) {
+        rtc::StringBuilder ss;
+        ss << "Checking event " << event << " at sample rate " << fs_hz;
+        ss << "; attenuation " << attenuation;
+        SCOPED_TRACE(ss.str());
+        ASSERT_EQ(0, tone_gen_.Init(fs_hz, event, attenuation));
+        EXPECT_EQ(kNumSamples, tone_gen_.Generate(kNumSamples, &signal));
+        for (int n = 0; n < kNumSamples; ++n) {
+          double attenuation_factor =
+              DbToRatio(-static_cast<float>(attenuation));
+          // Verify that the attenuation is correct.
+          for (int channel = 0; channel < channels; ++channel) {
+            EXPECT_NEAR(attenuation_factor * ref_signal[channel][n],
+                        signal[channel][n], 2);
+          }
+        }
+
+        tone_gen_.Reset();
+      }
+    }
+  }
+
+  DtmfToneGenerator tone_gen_;
+};
+
+// Low and high frequencies for events 0 through 15.
+const double DtmfToneGeneratorTest::kLowFreqHz[16] = {
+    941.0, 697.0, 697.0, 697.0, 770.0, 770.0, 770.0, 852.0,
+    852.0, 852.0, 941.0, 941.0, 697.0, 770.0, 852.0, 941.0};
+const double DtmfToneGeneratorTest::kHighFreqHz[16] = {
+    1336.0, 1209.0, 1336.0, 1477.0, 1209.0, 1336.0, 1477.0, 1209.0,
+    1336.0, 1477.0, 1209.0, 1477.0, 1633.0, 1633.0, 1633.0, 1633.0};
+
+TEST_F(DtmfToneGeneratorTest, Test8000Mono) {
+  TestAllTones(8000, 1);
+  TestAmplitudes(8000, 1);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test16000Mono) {
+  TestAllTones(16000, 1);
+  TestAmplitudes(16000, 1);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test32000Mono) {
+  TestAllTones(32000, 1);
+  TestAmplitudes(32000, 1);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test48000Mono) {
+  TestAllTones(48000, 1);
+  TestAmplitudes(48000, 1);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test8000Stereo) {
+  TestAllTones(8000, 2);
+  TestAmplitudes(8000, 2);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test16000Stereo) {
+  TestAllTones(16000, 2);
+  TestAmplitudes(16000, 2);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test32000Stereo) {
+  TestAllTones(32000, 2);
+  TestAmplitudes(32000, 2);
+}
+
+TEST_F(DtmfToneGeneratorTest, Test48000Stereo) {
+  TestAllTones(48000, 2);
+  TestAmplitudes(48000, 2);
+}
+
+TEST(DtmfToneGenerator, TestErrors) {
+  DtmfToneGenerator tone_gen;
+  const int kNumSamples = 10;
+  AudioMultiVector signal(1);  // One channel.
+
+  // Try to generate tones without initializing.
+  EXPECT_EQ(DtmfToneGenerator::kNotInitialized,
+            tone_gen.Generate(kNumSamples, &signal));
+
+  const int fs = 16000;       // Valid sample rate.
+  const int event = 7;        // Valid event.
+  const int attenuation = 0;  // Valid attenuation.
+  // Initialize with invalid event -1.
+  EXPECT_EQ(DtmfToneGenerator::kParameterError,
+            tone_gen.Init(fs, -1, attenuation));
+  // Initialize with invalid event 16.
+  EXPECT_EQ(DtmfToneGenerator::kParameterError,
+            tone_gen.Init(fs, 16, attenuation));
+  // Initialize with invalid attenuation -1.
+  EXPECT_EQ(DtmfToneGenerator::kParameterError, tone_gen.Init(fs, event, -1));
+  // Initialize with invalid attenuation 64.
+  EXPECT_EQ(DtmfToneGenerator::kParameterError, tone_gen.Init(fs, event, 64));
+  EXPECT_FALSE(tone_gen.initialized());  // Should still be uninitialized.
+
+  // Initialize with valid parameters.
+  ASSERT_EQ(0, tone_gen.Init(fs, event, attenuation));
+  EXPECT_TRUE(tone_gen.initialized());
+  // NULL pointer to destination.
+  EXPECT_EQ(DtmfToneGenerator::kParameterError,
+            tone_gen.Generate(kNumSamples, NULL));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/expand.cc b/third_party/libwebrtc/modules/audio_coding/neteq/expand.cc
new file mode 100644
index 0000000000..9c3274609f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/expand.cc
@@ -0,0 +1,888 @@
+/*
+ *  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/neteq/expand.h"
+
+#include <string.h>  // memset
+
+#include <algorithm>  // min, max
+#include <limits>     // numeric_limits<T>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+Expand::Expand(BackgroundNoise* background_noise,
+               SyncBuffer* sync_buffer,
+               RandomVector* random_vector,
+               StatisticsCalculator* statistics,
+               int fs,
+               size_t num_channels)
+    : random_vector_(random_vector),
+      sync_buffer_(sync_buffer),
+      first_expand_(true),
+      fs_hz_(fs),
+      num_channels_(num_channels),
+      consecutive_expands_(0),
+      background_noise_(background_noise),
+      statistics_(statistics),
+      overlap_length_(5 * fs / 8000),
+      lag_index_direction_(0),
+      current_lag_index_(0),
+      stop_muting_(false),
+      expand_duration_samples_(0),
+      channel_parameters_(new ChannelParameters[num_channels_]) {
+  RTC_DCHECK(fs == 8000 || fs == 16000 || fs == 32000 || fs == 48000);
+  RTC_DCHECK_LE(fs,
+                static_cast<int>(kMaxSampleRate));  // Should not be possible.
+  RTC_DCHECK_GT(num_channels_, 0);
+  memset(expand_lags_, 0, sizeof(expand_lags_));
+  Reset();
+}
+
+Expand::~Expand() = default;
+
+void Expand::Reset() {
+  first_expand_ = true;
+  consecutive_expands_ = 0;
+  max_lag_ = 0;
+  for (size_t ix = 0; ix < num_channels_; ++ix) {
+    channel_parameters_[ix].expand_vector0.Clear();
+    channel_parameters_[ix].expand_vector1.Clear();
+  }
+}
+
+int Expand::Process(AudioMultiVector* output) {
+  int16_t random_vector[kMaxSampleRate / 8000 * 120 + 30];
+  int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125];
+  static const int kTempDataSize = 3600;
+  int16_t temp_data[kTempDataSize];  // TODO(hlundin) Remove this.
+  int16_t* voiced_vector_storage = temp_data;
+  int16_t* voiced_vector = &voiced_vector_storage[overlap_length_];
+  static const size_t kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
+  int16_t unvoiced_array_memory[kNoiseLpcOrder + kMaxSampleRate / 8000 * 125];
+  int16_t* unvoiced_vector = unvoiced_array_memory + kUnvoicedLpcOrder;
+  int16_t* noise_vector = unvoiced_array_memory + kNoiseLpcOrder;
+
+  int fs_mult = fs_hz_ / 8000;
+
+  if (first_expand_) {
+    // Perform initial setup if this is the first expansion since last reset.
+    AnalyzeSignal(random_vector);
+    first_expand_ = false;
+    expand_duration_samples_ = 0;
+  } else {
+    // This is not the first expansion, parameters are already estimated.
+    // Extract a noise segment.
+    size_t rand_length = max_lag_;
+    // This only applies to SWB where length could be larger than 256.
+    RTC_DCHECK_LE(rand_length, kMaxSampleRate / 8000 * 120 + 30);
+    GenerateRandomVector(2, rand_length, random_vector);
+  }
+
+  // Generate signal.
+  UpdateLagIndex();
+
+  // Voiced part.
+  // Generate a weighted vector with the current lag.
+  size_t expansion_vector_length = max_lag_ + overlap_length_;
+  size_t current_lag = expand_lags_[current_lag_index_];
+  // Copy lag+overlap data.
+  size_t expansion_vector_position =
+      expansion_vector_length - current_lag - overlap_length_;
+  size_t temp_length = current_lag + overlap_length_;
+  for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
+    ChannelParameters& parameters = channel_parameters_[channel_ix];
+    if (current_lag_index_ == 0) {
+      // Use only expand_vector0.
+      RTC_DCHECK_LE(expansion_vector_position + temp_length,
+                    parameters.expand_vector0.Size());
+      parameters.expand_vector0.CopyTo(temp_length, expansion_vector_position,
+                                       voiced_vector_storage);
+    } else if (current_lag_index_ == 1) {
+      std::unique_ptr<int16_t[]> temp_0(new int16_t[temp_length]);
+      parameters.expand_vector0.CopyTo(temp_length, expansion_vector_position,
+                                       temp_0.get());
+      std::unique_ptr<int16_t[]> temp_1(new int16_t[temp_length]);
+      parameters.expand_vector1.CopyTo(temp_length, expansion_vector_position,
+                                       temp_1.get());
+      // Mix 3/4 of expand_vector0 with 1/4 of expand_vector1.
+      WebRtcSpl_ScaleAndAddVectorsWithRound(temp_0.get(), 3, temp_1.get(), 1, 2,
+                                            voiced_vector_storage, temp_length);
+    } else if (current_lag_index_ == 2) {
+      // Mix 1/2 of expand_vector0 with 1/2 of expand_vector1.
+      RTC_DCHECK_LE(expansion_vector_position + temp_length,
+                    parameters.expand_vector0.Size());
+      RTC_DCHECK_LE(expansion_vector_position + temp_length,
+                    parameters.expand_vector1.Size());
+
+      std::unique_ptr<int16_t[]> temp_0(new int16_t[temp_length]);
+      parameters.expand_vector0.CopyTo(temp_length, expansion_vector_position,
+                                       temp_0.get());
+      std::unique_ptr<int16_t[]> temp_1(new int16_t[temp_length]);
+      parameters.expand_vector1.CopyTo(temp_length, expansion_vector_position,
+                                       temp_1.get());
+      WebRtcSpl_ScaleAndAddVectorsWithRound(temp_0.get(), 1, temp_1.get(), 1, 1,
+                                            voiced_vector_storage, temp_length);
+    }
+
+    // Get tapering window parameters. Values are in Q15.
+    int16_t muting_window, muting_window_increment;
+    int16_t unmuting_window, unmuting_window_increment;
+    if (fs_hz_ == 8000) {
+      muting_window = DspHelper::kMuteFactorStart8kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement8kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart8kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement8kHz;
+    } else if (fs_hz_ == 16000) {
+      muting_window = DspHelper::kMuteFactorStart16kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement16kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart16kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement16kHz;
+    } else if (fs_hz_ == 32000) {
+      muting_window = DspHelper::kMuteFactorStart32kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement32kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart32kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement32kHz;
+    } else {  // fs_ == 48000
+      muting_window = DspHelper::kMuteFactorStart48kHz;
+      muting_window_increment = DspHelper::kMuteFactorIncrement48kHz;
+      unmuting_window = DspHelper::kUnmuteFactorStart48kHz;
+      unmuting_window_increment = DspHelper::kUnmuteFactorIncrement48kHz;
+    }
+
+    // Smooth the expanded if it has not been muted to a low amplitude and
+    // `current_voice_mix_factor` is larger than 0.5.
+    if ((parameters.mute_factor > 819) &&
+        (parameters.current_voice_mix_factor > 8192)) {
+      size_t start_ix = sync_buffer_->Size() - overlap_length_;
+      for (size_t i = 0; i < overlap_length_; i++) {
+        // Do overlap add between new vector and overlap.
+        (*sync_buffer_)[channel_ix][start_ix + i] =
+            (((*sync_buffer_)[channel_ix][start_ix + i] * muting_window) +
+             (((parameters.mute_factor * voiced_vector_storage[i]) >> 14) *
+              unmuting_window) +
+             16384) >>
+            15;
+        muting_window += muting_window_increment;
+        unmuting_window += unmuting_window_increment;
+      }
+    } else if (parameters.mute_factor == 0) {
+      // The expanded signal will consist of only comfort noise if
+      // mute_factor = 0. Set the output length to 15 ms for best noise
+      // production.
+      // TODO(hlundin): This has been disabled since the length of
+      // parameters.expand_vector0 and parameters.expand_vector1 no longer
+      // match with expand_lags_, causing invalid reads and writes. Is it a good
+      // idea to enable this again, and solve the vector size problem?
+      //      max_lag_ = fs_mult * 120;
+      //      expand_lags_[0] = fs_mult * 120;
+      //      expand_lags_[1] = fs_mult * 120;
+      //      expand_lags_[2] = fs_mult * 120;
+    }
+
+    // Unvoiced part.
+    // Filter `scaled_random_vector` through `ar_filter_`.
+    memcpy(unvoiced_vector - kUnvoicedLpcOrder, parameters.ar_filter_state,
+           sizeof(int16_t) * kUnvoicedLpcOrder);
+    int32_t add_constant = 0;
+    if (parameters.ar_gain_scale > 0) {
+      add_constant = 1 << (parameters.ar_gain_scale - 1);
+    }
+    WebRtcSpl_AffineTransformVector(scaled_random_vector, random_vector,
+                                    parameters.ar_gain, add_constant,
+                                    parameters.ar_gain_scale, current_lag);
+    WebRtcSpl_FilterARFastQ12(scaled_random_vector, unvoiced_vector,
+                              parameters.ar_filter, kUnvoicedLpcOrder + 1,
+                              current_lag);
+    memcpy(parameters.ar_filter_state,
+           &(unvoiced_vector[current_lag - kUnvoicedLpcOrder]),
+           sizeof(int16_t) * kUnvoicedLpcOrder);
+
+    // Combine voiced and unvoiced contributions.
+
+    // Set a suitable cross-fading slope.
+    // For lag =
+    //   <= 31 * fs_mult            => go from 1 to 0 in about 8 ms;
+    //  (>= 31 .. <= 63) * fs_mult  => go from 1 to 0 in about 16 ms;
+    //   >= 64 * fs_mult            => go from 1 to 0 in about 32 ms.
+    // temp_shift = getbits(max_lag_) - 5.
+    int temp_shift =
+        (31 - WebRtcSpl_NormW32(rtc::dchecked_cast<int32_t>(max_lag_))) - 5;
+    int16_t mix_factor_increment = 256 >> temp_shift;
+    if (stop_muting_) {
+      mix_factor_increment = 0;
+    }
+
+    // Create combined signal by shifting in more and more of unvoiced part.
+    temp_shift = 8 - temp_shift;  // = getbits(mix_factor_increment).
+    size_t temp_length =
+        (parameters.current_voice_mix_factor - parameters.voice_mix_factor) >>
+        temp_shift;
+    temp_length = std::min(temp_length, current_lag);
+    DspHelper::CrossFade(voiced_vector, unvoiced_vector, temp_length,
+                         &parameters.current_voice_mix_factor,
+                         mix_factor_increment, temp_data);
+
+    // End of cross-fading period was reached before end of expanded signal
+    // path. Mix the rest with a fixed mixing factor.
+    if (temp_length < current_lag) {
+      if (mix_factor_increment != 0) {
+        parameters.current_voice_mix_factor = parameters.voice_mix_factor;
+      }
+      int16_t temp_scale = 16384 - parameters.current_voice_mix_factor;
+      WebRtcSpl_ScaleAndAddVectorsWithRound(
+          voiced_vector + temp_length, parameters.current_voice_mix_factor,
+          unvoiced_vector + temp_length, temp_scale, 14,
+          temp_data + temp_length, current_lag - temp_length);
+    }
+
+    // Select muting slope depending on how many consecutive expands we have
+    // done.
+    if (consecutive_expands_ == 3) {
+      // Let the mute factor decrease from 1.0 to 0.95 in 6.25 ms.
+      // mute_slope = 0.0010 / fs_mult in Q20.
+      parameters.mute_slope = std::max(parameters.mute_slope, 1049 / fs_mult);
+    }
+    if (consecutive_expands_ == 7) {
+      // Let the mute factor decrease from 1.0 to 0.90 in 6.25 ms.
+      // mute_slope = 0.0020 / fs_mult in Q20.
+      parameters.mute_slope = std::max(parameters.mute_slope, 2097 / fs_mult);
+    }
+
+    // Mute segment according to slope value.
+    if ((consecutive_expands_ != 0) || !parameters.onset) {
+      // Mute to the previous level, then continue with the muting.
+      WebRtcSpl_AffineTransformVector(
+          temp_data, temp_data, parameters.mute_factor, 8192, 14, current_lag);
+
+      if (!stop_muting_) {
+        DspHelper::MuteSignal(temp_data, parameters.mute_slope, current_lag);
+
+        // Shift by 6 to go from Q20 to Q14.
+        // TODO(hlundin): Adding 8192 before shifting 6 steps seems wrong.
+        // Legacy.
+        int16_t gain = static_cast<int16_t>(
+            16384 - (((current_lag * parameters.mute_slope) + 8192) >> 6));
+        gain = ((gain * parameters.mute_factor) + 8192) >> 14;
+
+        // Guard against getting stuck with very small (but sometimes audible)
+        // gain.
+        if ((consecutive_expands_ > 3) && (gain >= parameters.mute_factor)) {
+          parameters.mute_factor = 0;
+        } else {
+          parameters.mute_factor = gain;
+        }
+      }
+    }
+
+    // Background noise part.
+    background_noise_->GenerateBackgroundNoise(
+        random_vector, channel_ix, channel_parameters_[channel_ix].mute_slope,
+        TooManyExpands(), current_lag, unvoiced_array_memory);
+
+    // Add background noise to the combined voiced-unvoiced signal.
+    for (size_t i = 0; i < current_lag; i++) {
+      temp_data[i] = temp_data[i] + noise_vector[i];
+    }
+    if (channel_ix == 0) {
+      output->AssertSize(current_lag);
+    } else {
+      RTC_DCHECK_EQ(output->Size(), current_lag);
+    }
+    (*output)[channel_ix].OverwriteAt(temp_data, current_lag, 0);
+  }
+
+  // Increase call number and cap it.
+  consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands
+                             ? kMaxConsecutiveExpands
+                             : consecutive_expands_ + 1;
+  expand_duration_samples_ += output->Size();
+  // Clamp the duration counter at 2 seconds.
+  expand_duration_samples_ = std::min(expand_duration_samples_,
+                                      rtc::dchecked_cast<size_t>(fs_hz_ * 2));
+  return 0;
+}
+
+void Expand::SetParametersForNormalAfterExpand() {
+  current_lag_index_ = 0;
+  lag_index_direction_ = 0;
+  stop_muting_ = true;  // Do not mute signal any more.
+  statistics_->LogDelayedPacketOutageEvent(expand_duration_samples_, fs_hz_);
+  statistics_->EndExpandEvent(fs_hz_);
+}
+
+void Expand::SetParametersForMergeAfterExpand() {
+  current_lag_index_ = -1;  /* out of the 3 possible ones */
+  lag_index_direction_ = 1; /* make sure we get the "optimal" lag */
+  stop_muting_ = true;
+  statistics_->EndExpandEvent(fs_hz_);
+}
+
+bool Expand::Muted() const {
+  if (first_expand_ || stop_muting_)
+    return false;
+  RTC_DCHECK(channel_parameters_);
+  for (size_t ch = 0; ch < num_channels_; ++ch) {
+    if (channel_parameters_[ch].mute_factor != 0)
+      return false;
+  }
+  return true;
+}
+
+size_t Expand::overlap_length() const {
+  return overlap_length_;
+}
+
+void Expand::InitializeForAnExpandPeriod() {
+  lag_index_direction_ = 1;
+  current_lag_index_ = -1;
+  stop_muting_ = false;
+  random_vector_->set_seed_increment(1);
+  consecutive_expands_ = 0;
+  for (size_t ix = 0; ix < num_channels_; ++ix) {
+    channel_parameters_[ix].current_voice_mix_factor = 16384;  // 1.0 in Q14.
+    channel_parameters_[ix].mute_factor = 16384;               // 1.0 in Q14.
+    // Start with 0 gain for background noise.
+    background_noise_->SetMuteFactor(ix, 0);
+  }
+}
+
+bool Expand::TooManyExpands() {
+  return consecutive_expands_ >= kMaxConsecutiveExpands;
+}
+
+void Expand::AnalyzeSignal(int16_t* random_vector) {
+  int32_t auto_correlation[kUnvoicedLpcOrder + 1];
+  int16_t reflection_coeff[kUnvoicedLpcOrder];
+  int16_t correlation_vector[kMaxSampleRate / 8000 * 102];
+  size_t best_correlation_index[kNumCorrelationCandidates];
+  int16_t best_correlation[kNumCorrelationCandidates];
+  size_t best_distortion_index[kNumCorrelationCandidates];
+  int16_t best_distortion[kNumCorrelationCandidates];
+  int32_t correlation_vector2[(99 * kMaxSampleRate / 8000) + 1];
+  int32_t best_distortion_w32[kNumCorrelationCandidates];
+  static const size_t kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
+  int16_t unvoiced_array_memory[kNoiseLpcOrder + kMaxSampleRate / 8000 * 125];
+  int16_t* unvoiced_vector = unvoiced_array_memory + kUnvoicedLpcOrder;
+
+  int fs_mult = fs_hz_ / 8000;
+
+  // Pre-calculate common multiplications with fs_mult.
+  size_t fs_mult_4 = static_cast<size_t>(fs_mult * 4);
+  size_t fs_mult_20 = static_cast<size_t>(fs_mult * 20);
+  size_t fs_mult_120 = static_cast<size_t>(fs_mult * 120);
+  size_t fs_mult_dist_len = fs_mult * kDistortionLength;
+  size_t fs_mult_lpc_analysis_len = fs_mult * kLpcAnalysisLength;
+
+  const size_t signal_length = static_cast<size_t>(256 * fs_mult);
+
+  const size_t audio_history_position = sync_buffer_->Size() - signal_length;
+  std::unique_ptr<int16_t[]> audio_history(new int16_t[signal_length]);
+  (*sync_buffer_)[0].CopyTo(signal_length, audio_history_position,
+                            audio_history.get());
+
+  // Initialize.
+  InitializeForAnExpandPeriod();
+
+  // Calculate correlation in downsampled domain (4 kHz sample rate).
+  size_t correlation_length = 51;  // TODO(hlundin): Legacy bit-exactness.
+  // If it is decided to break bit-exactness `correlation_length` should be
+  // initialized to the return value of Correlation().
+  Correlation(audio_history.get(), signal_length, correlation_vector);
+
+  // Find peaks in correlation vector.
+  DspHelper::PeakDetection(correlation_vector, correlation_length,
+                           kNumCorrelationCandidates, fs_mult,
+                           best_correlation_index, best_correlation);
+
+  // Adjust peak locations; cross-correlation lags start at 2.5 ms
+  // (20 * fs_mult samples).
+  best_correlation_index[0] += fs_mult_20;
+  best_correlation_index[1] += fs_mult_20;
+  best_correlation_index[2] += fs_mult_20;
+
+  // Calculate distortion around the `kNumCorrelationCandidates` best lags.
+  int distortion_scale = 0;
+  for (size_t i = 0; i < kNumCorrelationCandidates; i++) {
+    size_t min_index =
+        std::max(fs_mult_20, best_correlation_index[i] - fs_mult_4);
+    size_t max_index =
+        std::min(fs_mult_120 - 1, best_correlation_index[i] + fs_mult_4);
+    best_distortion_index[i] = DspHelper::MinDistortion(
+        &(audio_history[signal_length - fs_mult_dist_len]), min_index,
+        max_index, fs_mult_dist_len, &best_distortion_w32[i]);
+    distortion_scale = std::max(16 - WebRtcSpl_NormW32(best_distortion_w32[i]),
+                                distortion_scale);
+  }
+  // Shift the distortion values to fit in 16 bits.
+  WebRtcSpl_VectorBitShiftW32ToW16(best_distortion, kNumCorrelationCandidates,
+                                   best_distortion_w32, distortion_scale);
+
+  // Find the maximizing index `i` of the cost function
+  // f[i] = best_correlation[i] / best_distortion[i].
+  int32_t best_ratio = std::numeric_limits<int32_t>::min();
+  size_t best_index = std::numeric_limits<size_t>::max();
+  for (size_t i = 0; i < kNumCorrelationCandidates; ++i) {
+    int32_t ratio;
+    if (best_distortion[i] > 0) {
+      ratio = (best_correlation[i] * (1 << 16)) / best_distortion[i];
+    } else if (best_correlation[i] == 0) {
+      ratio = 0;  // No correlation set result to zero.
+    } else {
+      ratio = std::numeric_limits<int32_t>::max();  // Denominator is zero.
+    }
+    if (ratio > best_ratio) {
+      best_index = i;
+      best_ratio = ratio;
+    }
+  }
+
+  size_t distortion_lag = best_distortion_index[best_index];
+  size_t correlation_lag = best_correlation_index[best_index];
+  max_lag_ = std::max(distortion_lag, correlation_lag);
+
+  // Calculate the exact best correlation in the range between
+  // `correlation_lag` and `distortion_lag`.
+  correlation_length = std::max(std::min(distortion_lag + 10, fs_mult_120),
+                                static_cast<size_t>(60 * fs_mult));
+
+  size_t start_index = std::min(distortion_lag, correlation_lag);
+  size_t correlation_lags = static_cast<size_t>(
+      WEBRTC_SPL_ABS_W16((distortion_lag - correlation_lag)) + 1);
+  RTC_DCHECK_LE(correlation_lags, static_cast<size_t>(99 * fs_mult + 1));
+
+  for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
+    ChannelParameters& parameters = channel_parameters_[channel_ix];
+    if (channel_ix > 0) {
+      // When channel_ix == 0, audio_history contains the correct audio. For the
+      // other cases, we will have to copy the correct channel into
+      // audio_history.
+      (*sync_buffer_)[channel_ix].CopyTo(signal_length, audio_history_position,
+                                         audio_history.get());
+    }
+
+    // Calculate suitable scaling.
+    int16_t signal_max = WebRtcSpl_MaxAbsValueW16(
+        &audio_history[signal_length - correlation_length - start_index -
+                       correlation_lags],
+        correlation_length + start_index + correlation_lags - 1);
+    int correlation_scale =
+        (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
+        (31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31;
+    correlation_scale = std::max(0, correlation_scale);
+
+    // Calculate the correlation, store in `correlation_vector2`.
+    WebRtcSpl_CrossCorrelation(
+        correlation_vector2,
+        &(audio_history[signal_length - correlation_length]),
+        &(audio_history[signal_length - correlation_length - start_index]),
+        correlation_length, correlation_lags, correlation_scale, -1);
+
+    // Find maximizing index.
+    best_index = WebRtcSpl_MaxIndexW32(correlation_vector2, correlation_lags);
+    int32_t max_correlation = correlation_vector2[best_index];
+    // Compensate index with start offset.
+    best_index = best_index + start_index;
+
+    // Calculate energies.
+    int32_t energy1 = WebRtcSpl_DotProductWithScale(
+        &(audio_history[signal_length - correlation_length]),
+        &(audio_history[signal_length - correlation_length]),
+        correlation_length, correlation_scale);
+    int32_t energy2 = WebRtcSpl_DotProductWithScale(
+        &(audio_history[signal_length - correlation_length - best_index]),
+        &(audio_history[signal_length - correlation_length - best_index]),
+        correlation_length, correlation_scale);
+
+    // Calculate the correlation coefficient between the two portions of the
+    // signal.
+    int32_t corr_coefficient;
+    if ((energy1 > 0) && (energy2 > 0)) {
+      int energy1_scale = std::max(16 - WebRtcSpl_NormW32(energy1), 0);
+      int energy2_scale = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
+      // Make sure total scaling is even (to simplify scale factor after sqrt).
+      if ((energy1_scale + energy2_scale) & 1) {
+        // If sum is odd, add 1 to make it even.
+        energy1_scale += 1;
+      }
+      int32_t scaled_energy1 = energy1 >> energy1_scale;
+      int32_t scaled_energy2 = energy2 >> energy2_scale;
+      int16_t sqrt_energy_product = static_cast<int16_t>(
+          WebRtcSpl_SqrtFloor(scaled_energy1 * scaled_energy2));
+      // Calculate max_correlation / sqrt(energy1 * energy2) in Q14.
+      int cc_shift = 14 - (energy1_scale + energy2_scale) / 2;
+      max_correlation = WEBRTC_SPL_SHIFT_W32(max_correlation, cc_shift);
+      corr_coefficient =
+          WebRtcSpl_DivW32W16(max_correlation, sqrt_energy_product);
+      // Cap at 1.0 in Q14.
+      corr_coefficient = std::min(16384, corr_coefficient);
+    } else {
+      corr_coefficient = 0;
+    }
+
+    // Extract the two vectors expand_vector0 and expand_vector1 from
+    // `audio_history`.
+    size_t expansion_length = max_lag_ + overlap_length_;
+    const int16_t* vector1 = &(audio_history[signal_length - expansion_length]);
+    const int16_t* vector2 = vector1 - distortion_lag;
+    // Normalize the second vector to the same energy as the first.
+    energy1 = WebRtcSpl_DotProductWithScale(vector1, vector1, expansion_length,
+                                            correlation_scale);
+    energy2 = WebRtcSpl_DotProductWithScale(vector2, vector2, expansion_length,
+                                            correlation_scale);
+    // Confirm that amplitude ratio sqrt(energy1 / energy2) is within 0.5 - 2.0,
+    // i.e., energy1 / energy2 is within 0.25 - 4.
+    int16_t amplitude_ratio;
+    if ((energy1 / 4 < energy2) && (energy1 > energy2 / 4)) {
+      // Energy constraint fulfilled. Use both vectors and scale them
+      // accordingly.
+      int32_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
+      int32_t scaled_energy1 = scaled_energy2 - 13;
+      // Calculate scaled_energy1 / scaled_energy2 in Q13.
+      int32_t energy_ratio =
+          WebRtcSpl_DivW32W16(WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
+                              static_cast<int16_t>(energy2 >> scaled_energy2));
+      // Calculate sqrt ratio in Q13 (sqrt of en1/en2 in Q26).
+      amplitude_ratio =
+          static_cast<int16_t>(WebRtcSpl_SqrtFloor(energy_ratio << 13));
+      // Copy the two vectors and give them the same energy.
+      parameters.expand_vector0.Clear();
+      parameters.expand_vector0.PushBack(vector1, expansion_length);
+      parameters.expand_vector1.Clear();
+      if (parameters.expand_vector1.Size() < expansion_length) {
+        parameters.expand_vector1.Extend(expansion_length -
+                                         parameters.expand_vector1.Size());
+      }
+      std::unique_ptr<int16_t[]> temp_1(new int16_t[expansion_length]);
+      WebRtcSpl_AffineTransformVector(
+          temp_1.get(), const_cast<int16_t*>(vector2), amplitude_ratio, 4096,
+          13, expansion_length);
+      parameters.expand_vector1.OverwriteAt(temp_1.get(), expansion_length, 0);
+    } else {
+      // Energy change constraint not fulfilled. Only use last vector.
+      parameters.expand_vector0.Clear();
+      parameters.expand_vector0.PushBack(vector1, expansion_length);
+      // Copy from expand_vector0 to expand_vector1.
+      parameters.expand_vector0.CopyTo(&parameters.expand_vector1);
+      // Set the energy_ratio since it is used by muting slope.
+      if ((energy1 / 4 < energy2) || (energy2 == 0)) {
+        amplitude_ratio = 4096;  // 0.5 in Q13.
+      } else {
+        amplitude_ratio = 16384;  // 2.0 in Q13.
+      }
+    }
+
+    // Set the 3 lag values.
+    if (distortion_lag == correlation_lag) {
+      expand_lags_[0] = distortion_lag;
+      expand_lags_[1] = distortion_lag;
+      expand_lags_[2] = distortion_lag;
+    } else {
+      // `distortion_lag` and `correlation_lag` are not equal; use different
+      // combinations of the two.
+      // First lag is `distortion_lag` only.
+      expand_lags_[0] = distortion_lag;
+      // Second lag is the average of the two.
+      expand_lags_[1] = (distortion_lag + correlation_lag) / 2;
+      // Third lag is the average again, but rounding towards `correlation_lag`.
+      if (distortion_lag > correlation_lag) {
+        expand_lags_[2] = (distortion_lag + correlation_lag - 1) / 2;
+      } else {
+        expand_lags_[2] = (distortion_lag + correlation_lag + 1) / 2;
+      }
+    }
+
+    // Calculate the LPC and the gain of the filters.
+
+    // Calculate kUnvoicedLpcOrder + 1 lags of the auto-correlation function.
+    size_t temp_index =
+        signal_length - fs_mult_lpc_analysis_len - kUnvoicedLpcOrder;
+    // Copy signal to temporary vector to be able to pad with leading zeros.
+    int16_t* temp_signal =
+        new int16_t[fs_mult_lpc_analysis_len + kUnvoicedLpcOrder];
+    memset(temp_signal, 0,
+           sizeof(int16_t) * (fs_mult_lpc_analysis_len + kUnvoicedLpcOrder));
+    memcpy(&temp_signal[kUnvoicedLpcOrder],
+           &audio_history[temp_index + kUnvoicedLpcOrder],
+           sizeof(int16_t) * fs_mult_lpc_analysis_len);
+    CrossCorrelationWithAutoShift(
+        &temp_signal[kUnvoicedLpcOrder], &temp_signal[kUnvoicedLpcOrder],
+        fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1, -1, auto_correlation);
+    delete[] temp_signal;
+
+    // Verify that variance is positive.
+    if (auto_correlation[0] > 0) {
+      // Estimate AR filter parameters using Levinson-Durbin algorithm;
+      // kUnvoicedLpcOrder + 1 filter coefficients.
+      int16_t stability =
+          WebRtcSpl_LevinsonDurbin(auto_correlation, parameters.ar_filter,
+                                   reflection_coeff, kUnvoicedLpcOrder);
+
+      // Keep filter parameters only if filter is stable.
+      if (stability != 1) {
+        // Set first coefficient to 4096 (1.0 in Q12).
+        parameters.ar_filter[0] = 4096;
+        // Set remaining `kUnvoicedLpcOrder` coefficients to zero.
+        WebRtcSpl_MemSetW16(parameters.ar_filter + 1, 0, kUnvoicedLpcOrder);
+      }
+    }
+
+    if (channel_ix == 0) {
+      // Extract a noise segment.
+      size_t noise_length;
+      if (distortion_lag < 40) {
+        noise_length = 2 * distortion_lag + 30;
+      } else {
+        noise_length = distortion_lag + 30;
+      }
+      if (noise_length <= RandomVector::kRandomTableSize) {
+        memcpy(random_vector, RandomVector::kRandomTable,
+               sizeof(int16_t) * noise_length);
+      } else {
+        // Only applies to SWB where length could be larger than
+        // `kRandomTableSize`.
+        memcpy(random_vector, RandomVector::kRandomTable,
+               sizeof(int16_t) * RandomVector::kRandomTableSize);
+        RTC_DCHECK_LE(noise_length, kMaxSampleRate / 8000 * 120 + 30);
+        random_vector_->IncreaseSeedIncrement(2);
+        random_vector_->Generate(
+            noise_length - RandomVector::kRandomTableSize,
+            &random_vector[RandomVector::kRandomTableSize]);
+      }
+    }
+
+    // Set up state vector and calculate scale factor for unvoiced filtering.
+    memcpy(parameters.ar_filter_state,
+           &(audio_history[signal_length - kUnvoicedLpcOrder]),
+           sizeof(int16_t) * kUnvoicedLpcOrder);
+    memcpy(unvoiced_vector - kUnvoicedLpcOrder,
+           &(audio_history[signal_length - 128 - kUnvoicedLpcOrder]),
+           sizeof(int16_t) * kUnvoicedLpcOrder);
+    WebRtcSpl_FilterMAFastQ12(&audio_history[signal_length - 128],
+                              unvoiced_vector, parameters.ar_filter,
+                              kUnvoicedLpcOrder + 1, 128);
+    const int unvoiced_max_abs = [&] {
+      const int16_t max_abs = WebRtcSpl_MaxAbsValueW16(unvoiced_vector, 128);
+      // Since WebRtcSpl_MaxAbsValueW16 returns 2^15 - 1 when the input contains
+      // -2^15, we have to conservatively bump the return value by 1
+      // if it is 2^15 - 1.
+      return max_abs == WEBRTC_SPL_WORD16_MAX ? max_abs + 1 : max_abs;
+    }();
+    // Pick the smallest n such that 2^n > unvoiced_max_abs; then the maximum
+    // value of the dot product is less than 2^7 * 2^(2*n) = 2^(2*n + 7), so to
+    // prevent overflows we want 2n + 7 <= 31, which means we should shift by
+    // 2n + 7 - 31 bits, if this value is greater than zero.
+    int unvoiced_prescale =
+        std::max(0, 2 * WebRtcSpl_GetSizeInBits(unvoiced_max_abs) - 24);
+
+    int32_t unvoiced_energy = WebRtcSpl_DotProductWithScale(
+        unvoiced_vector, unvoiced_vector, 128, unvoiced_prescale);
+
+    // Normalize `unvoiced_energy` to 28 or 29 bits to preserve sqrt() accuracy.
+    int16_t unvoiced_scale = WebRtcSpl_NormW32(unvoiced_energy) - 3;
+    // Make sure we do an odd number of shifts since we already have 7 shifts
+    // from dividing with 128 earlier. This will make the total scale factor
+    // even, which is suitable for the sqrt.
+    unvoiced_scale += ((unvoiced_scale & 0x1) ^ 0x1);
+    unvoiced_energy = WEBRTC_SPL_SHIFT_W32(unvoiced_energy, unvoiced_scale);
+    int16_t unvoiced_gain =
+        static_cast<int16_t>(WebRtcSpl_SqrtFloor(unvoiced_energy));
+    parameters.ar_gain_scale =
+        13 + (unvoiced_scale + 7 - unvoiced_prescale) / 2;
+    parameters.ar_gain = unvoiced_gain;
+
+    // Calculate voice_mix_factor from corr_coefficient.
+    // Let x = corr_coefficient. Then, we compute:
+    // if (x > 0.48)
+    //   voice_mix_factor = (-5179 + 19931x - 16422x^2 + 5776x^3) / 4096;
+    // else
+    //   voice_mix_factor = 0;
+    if (corr_coefficient > 7875) {
+      int16_t x1, x2, x3;
+      // `corr_coefficient` is in Q14.
+      x1 = static_cast<int16_t>(corr_coefficient);
+      x2 = (x1 * x1) >> 14;  // Shift 14 to keep result in Q14.
+      x3 = (x1 * x2) >> 14;
+      static const int kCoefficients[4] = {-5179, 19931, -16422, 5776};
+      int32_t temp_sum = kCoefficients[0] * 16384;
+      temp_sum += kCoefficients[1] * x1;
+      temp_sum += kCoefficients[2] * x2;
+      temp_sum += kCoefficients[3] * x3;
+      parameters.voice_mix_factor =
+          static_cast<int16_t>(std::min(temp_sum / 4096, 16384));
+      parameters.voice_mix_factor =
+          std::max(parameters.voice_mix_factor, static_cast<int16_t>(0));
+    } else {
+      parameters.voice_mix_factor = 0;
+    }
+
+    // Calculate muting slope. Reuse value from earlier scaling of
+    // `expand_vector0` and `expand_vector1`.
+    int16_t slope = amplitude_ratio;
+    if (slope > 12288) {
+      // slope > 1.5.
+      // Calculate (1 - (1 / slope)) / distortion_lag =
+      // (slope - 1) / (distortion_lag * slope).
+      // `slope` is in Q13, so 1 corresponds to 8192. Shift up to Q25 before
+      // the division.
+      // Shift the denominator from Q13 to Q5 before the division. The result of
+      // the division will then be in Q20.
+      int16_t denom =
+          rtc::saturated_cast<int16_t>((distortion_lag * slope) >> 8);
+      int temp_ratio = WebRtcSpl_DivW32W16((slope - 8192) << 12, denom);
+      if (slope > 14746) {
+        // slope > 1.8.
+        // Divide by 2, with proper rounding.
+        parameters.mute_slope = (temp_ratio + 1) / 2;
+      } else {
+        // Divide by 8, with proper rounding.
+        parameters.mute_slope = (temp_ratio + 4) / 8;
+      }
+      parameters.onset = true;
+    } else {
+      // Calculate (1 - slope) / distortion_lag.
+      // Shift `slope` by 7 to Q20 before the division. The result is in Q20.
+      parameters.mute_slope = WebRtcSpl_DivW32W16(
+          (8192 - slope) * 128, static_cast<int16_t>(distortion_lag));
+      if (parameters.voice_mix_factor <= 13107) {
+        // Make sure the mute factor decreases from 1.0 to 0.9 in no more than
+        // 6.25 ms.
+        // mute_slope >= 0.005 / fs_mult in Q20.
+        parameters.mute_slope = std::max(5243 / fs_mult, parameters.mute_slope);
+      } else if (slope > 8028) {
+        parameters.mute_slope = 0;
+      }
+      parameters.onset = false;
+    }
+  }
+}
+
+Expand::ChannelParameters::ChannelParameters()
+    : mute_factor(16384),
+      ar_gain(0),
+      ar_gain_scale(0),
+      voice_mix_factor(0),
+      current_voice_mix_factor(0),
+      onset(false),
+      mute_slope(0) {
+  memset(ar_filter, 0, sizeof(ar_filter));
+  memset(ar_filter_state, 0, sizeof(ar_filter_state));
+}
+
+void Expand::Correlation(const int16_t* input,
+                         size_t input_length,
+                         int16_t* output) const {
+  // Set parameters depending on sample rate.
+  const int16_t* filter_coefficients;
+  size_t num_coefficients;
+  int16_t downsampling_factor;
+  if (fs_hz_ == 8000) {
+    num_coefficients = 3;
+    downsampling_factor = 2;
+    filter_coefficients = DspHelper::kDownsample8kHzTbl;
+  } else if (fs_hz_ == 16000) {
+    num_coefficients = 5;
+    downsampling_factor = 4;
+    filter_coefficients = DspHelper::kDownsample16kHzTbl;
+  } else if (fs_hz_ == 32000) {
+    num_coefficients = 7;
+    downsampling_factor = 8;
+    filter_coefficients = DspHelper::kDownsample32kHzTbl;
+  } else {  // fs_hz_ == 48000.
+    num_coefficients = 7;
+    downsampling_factor = 12;
+    filter_coefficients = DspHelper::kDownsample48kHzTbl;
+  }
+
+  // Correlate from lag 10 to lag 60 in downsampled domain.
+  // (Corresponds to 20-120 for narrow-band, 40-240 for wide-band, and so on.)
+  static const size_t kCorrelationStartLag = 10;
+  static const size_t kNumCorrelationLags = 54;
+  static const size_t kCorrelationLength = 60;
+  // Downsample to 4 kHz sample rate.
+  static const size_t kDownsampledLength =
+      kCorrelationStartLag + kNumCorrelationLags + kCorrelationLength;
+  int16_t downsampled_input[kDownsampledLength];
+  static const size_t kFilterDelay = 0;
+  WebRtcSpl_DownsampleFast(
+      input + input_length - kDownsampledLength * downsampling_factor,
+      kDownsampledLength * downsampling_factor, downsampled_input,
+      kDownsampledLength, filter_coefficients, num_coefficients,
+      downsampling_factor, kFilterDelay);
+
+  // Normalize `downsampled_input` to using all 16 bits.
+  int16_t max_value =
+      WebRtcSpl_MaxAbsValueW16(downsampled_input, kDownsampledLength);
+  int16_t norm_shift = 16 - WebRtcSpl_NormW32(max_value);
+  WebRtcSpl_VectorBitShiftW16(downsampled_input, kDownsampledLength,
+                              downsampled_input, norm_shift);
+
+  int32_t correlation[kNumCorrelationLags];
+  CrossCorrelationWithAutoShift(
+      &downsampled_input[kDownsampledLength - kCorrelationLength],
+      &downsampled_input[kDownsampledLength - kCorrelationLength -
+                         kCorrelationStartLag],
+      kCorrelationLength, kNumCorrelationLags, -1, correlation);
+
+  // Normalize and move data from 32-bit to 16-bit vector.
+  int32_t max_correlation =
+      WebRtcSpl_MaxAbsValueW32(correlation, kNumCorrelationLags);
+  int16_t norm_shift2 = static_cast<int16_t>(
+      std::max(18 - WebRtcSpl_NormW32(max_correlation), 0));
+  WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation,
+                                   norm_shift2);
+}
+
+void Expand::UpdateLagIndex() {
+  current_lag_index_ = current_lag_index_ + lag_index_direction_;
+  // Change direction if needed.
+  if (current_lag_index_ <= 0) {
+    lag_index_direction_ = 1;
+  }
+  if (current_lag_index_ >= kNumLags - 1) {
+    lag_index_direction_ = -1;
+  }
+}
+
+Expand* ExpandFactory::Create(BackgroundNoise* background_noise,
+                              SyncBuffer* sync_buffer,
+                              RandomVector* random_vector,
+                              StatisticsCalculator* statistics,
+                              int fs,
+                              size_t num_channels) const {
+  return new Expand(background_noise, sync_buffer, random_vector, statistics,
+                    fs, num_channels);
+}
+
+void Expand::GenerateRandomVector(int16_t seed_increment,
+                                  size_t length,
+                                  int16_t* random_vector) {
+  // TODO(turajs): According to hlundin The loop should not be needed. Should be
+  // just as good to generate all of the vector in one call.
+  size_t samples_generated = 0;
+  const size_t kMaxRandSamples = RandomVector::kRandomTableSize;
+  while (samples_generated < length) {
+    size_t rand_length = std::min(length - samples_generated, kMaxRandSamples);
+    random_vector_->IncreaseSeedIncrement(seed_increment);
+    random_vector_->Generate(rand_length, &random_vector[samples_generated]);
+    samples_generated += rand_length;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/expand.h b/third_party/libwebrtc/modules/audio_coding/neteq/expand.h
new file mode 100644
index 0000000000..2e64583ec2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/expand.h
@@ -0,0 +1,154 @@
+/*
+ *  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_NETEQ_EXPAND_H_
+#define MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
+
+
+#include <memory>
+
+#include "modules/audio_coding/neteq/audio_vector.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class AudioMultiVector;
+class BackgroundNoise;
+class RandomVector;
+class StatisticsCalculator;
+class SyncBuffer;
+
+// This class handles extrapolation of audio data from the sync_buffer to
+// produce packet-loss concealment.
+// TODO(hlundin): Refactor this class to divide the long methods into shorter
+// ones.
+class Expand {
+ public:
+  Expand(BackgroundNoise* background_noise,
+         SyncBuffer* sync_buffer,
+         RandomVector* random_vector,
+         StatisticsCalculator* statistics,
+         int fs,
+         size_t num_channels);
+
+  virtual ~Expand();
+
+  Expand(const Expand&) = delete;
+  Expand& operator=(const Expand&) = delete;
+
+  // Resets the object.
+  virtual void Reset();
+
+  // The main method to produce concealment data. The data is appended to the
+  // end of `output`.
+  virtual int Process(AudioMultiVector* output);
+
+  // Prepare the object to do extra expansion during normal operation following
+  // a period of expands.
+  virtual void SetParametersForNormalAfterExpand();
+
+  // Prepare the object to do extra expansion during merge operation following
+  // a period of expands.
+  virtual void SetParametersForMergeAfterExpand();
+
+  // Returns the mute factor for `channel`.
+  int16_t MuteFactor(size_t channel) const {
+    RTC_DCHECK_LT(channel, num_channels_);
+    return channel_parameters_[channel].mute_factor;
+  }
+
+  // Returns true if expansion has been faded down to zero amplitude (for all
+  // channels); false otherwise.
+  bool Muted() const;
+
+  // Accessors and mutators.
+  virtual size_t overlap_length() const;
+  size_t max_lag() const { return max_lag_; }
+
+ protected:
+  static const int kMaxConsecutiveExpands = 200;
+  void GenerateRandomVector(int16_t seed_increment,
+                            size_t length,
+                            int16_t* random_vector);
+
+  // Initializes member variables at the beginning of an expand period.
+  void InitializeForAnExpandPeriod();
+
+  bool TooManyExpands();
+
+  // Analyzes the signal history in `sync_buffer_`, and set up all parameters
+  // necessary to produce concealment data.
+  void AnalyzeSignal(int16_t* random_vector);
+
+  RandomVector* const random_vector_;
+  SyncBuffer* const sync_buffer_;
+  bool first_expand_;
+  const int fs_hz_;
+  const size_t num_channels_;
+  int consecutive_expands_;
+
+ private:
+  static const size_t kUnvoicedLpcOrder = 6;
+  static const size_t kNumCorrelationCandidates = 3;
+  static const size_t kDistortionLength = 20;
+  static const size_t kLpcAnalysisLength = 160;
+  static const size_t kMaxSampleRate = 48000;
+  static const int kNumLags = 3;
+
+  struct ChannelParameters {
+    ChannelParameters();
+    int16_t mute_factor;
+    int16_t ar_filter[kUnvoicedLpcOrder + 1];
+    int16_t ar_filter_state[kUnvoicedLpcOrder];
+    int16_t ar_gain;
+    int16_t ar_gain_scale;
+    int16_t voice_mix_factor;         /* Q14 */
+    int16_t current_voice_mix_factor; /* Q14 */
+    AudioVector expand_vector0;
+    AudioVector expand_vector1;
+    bool onset;
+    int mute_slope; /* Q20 */
+  };
+
+  // Calculate the auto-correlation of `input`, with length `input_length`
+  // samples. The correlation is calculated from a downsampled version of
+  // `input`, and is written to `output`.
+  void Correlation(const int16_t* input,
+                   size_t input_length,
+                   int16_t* output) const;
+
+  void UpdateLagIndex();
+
+  BackgroundNoise* const background_noise_;
+  StatisticsCalculator* const statistics_;
+  const size_t overlap_length_;
+  size_t max_lag_;
+  size_t expand_lags_[kNumLags];
+  int lag_index_direction_;
+  int current_lag_index_;
+  bool stop_muting_;
+  size_t expand_duration_samples_;
+  std::unique_ptr<ChannelParameters[]> channel_parameters_;
+};
+
+struct ExpandFactory {
+  ExpandFactory() {}
+  virtual ~ExpandFactory() {}
+
+  virtual Expand* Create(BackgroundNoise* background_noise,
+                         SyncBuffer* sync_buffer,
+                         RandomVector* random_vector,
+                         StatisticsCalculator* statistics,
+                         int fs,
+                         size_t num_channels) const;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.cc b/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.cc
new file mode 100644
index 0000000000..a91358b489
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.cc
@@ -0,0 +1,71 @@
+/*  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/neteq/expand_uma_logger.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+std::unique_ptr<TickTimer::Countdown> GetNewCountdown(
+    const TickTimer& tick_timer,
+    int logging_period_s) {
+  return tick_timer.GetNewCountdown((logging_period_s * 1000) /
+                                    tick_timer.ms_per_tick());
+}
+}  // namespace
+
+ExpandUmaLogger::ExpandUmaLogger(absl::string_view uma_name,
+                                 int logging_period_s,
+                                 const TickTimer* tick_timer)
+    : uma_name_(uma_name),
+      logging_period_s_(logging_period_s),
+      tick_timer_(*tick_timer),
+      timer_(GetNewCountdown(tick_timer_, logging_period_s_)) {
+  RTC_DCHECK(tick_timer);
+  RTC_DCHECK_GT(logging_period_s_, 0);
+}
+
+ExpandUmaLogger::~ExpandUmaLogger() = default;
+
+void ExpandUmaLogger::UpdateSampleCounter(uint64_t samples,
+                                          int sample_rate_hz) {
+  if ((last_logged_value_ && *last_logged_value_ > samples) ||
+      sample_rate_hz_ != sample_rate_hz) {
+    // Sanity checks. The incremental counter moved backwards, or sample rate
+    // changed.
+    last_logged_value_.reset();
+  }
+  last_value_ = samples;
+  sample_rate_hz_ = sample_rate_hz;
+  if (!last_logged_value_) {
+    last_logged_value_ = absl::optional<uint64_t>(samples);
+  }
+
+  if (!timer_->Finished()) {
+    // Not yet time to log.
+    return;
+  }
+
+  RTC_DCHECK(last_logged_value_);
+  RTC_DCHECK_GE(last_value_, *last_logged_value_);
+  const uint64_t diff = last_value_ - *last_logged_value_;
+  last_logged_value_ = absl::optional<uint64_t>(last_value_);
+  // Calculate rate in percent.
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  const int rate = (100 * diff) / (sample_rate_hz * logging_period_s_);
+  RTC_DCHECK_GE(rate, 0);
+  RTC_DCHECK_LE(rate, 100);
+  RTC_HISTOGRAM_PERCENTAGE_SPARSE(uma_name_, rate);
+  timer_ = GetNewCountdown(tick_timer_, logging_period_s_);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.h b/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.h
new file mode 100644
index 0000000000..cc5c20a886
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/expand_uma_logger.h
@@ -0,0 +1,57 @@
+/*  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_NETEQ_EXPAND_UMA_LOGGER_H_
+#define MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
+
+namespace webrtc {
+
+// This class is used to periodically log values to a UMA histogram. The caller
+// is expected to update this class with an incremental sample counter which
+// counts expand samples. At the end of each logging period, the class will
+// calculate the fraction of samples that were expand samples during that period
+// and report that in percent. The logging period must be strictly positive.
+// Does not take ownership of tick_timer and the pointer must refer to a valid
+// object that outlives the one constructed.
+class ExpandUmaLogger {
+ public:
+  ExpandUmaLogger(absl::string_view uma_name,
+                  int logging_period_s,
+                  const TickTimer* tick_timer);
+
+  ~ExpandUmaLogger();
+
+  ExpandUmaLogger(const ExpandUmaLogger&) = delete;
+  ExpandUmaLogger& operator=(const ExpandUmaLogger&) = delete;
+
+  // In this call, value should be an incremental sample counter. The sample
+  // rate must be strictly positive.
+  void UpdateSampleCounter(uint64_t value, int sample_rate_hz);
+
+ private:
+  const std::string uma_name_;
+  const int logging_period_s_;
+  const TickTimer& tick_timer_;
+  std::unique_ptr<TickTimer::Countdown> timer_;
+  absl::optional<uint64_t> last_logged_value_;
+  uint64_t last_value_ = 0;
+  int sample_rate_hz_ = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/expand_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/expand_unittest.cc
new file mode 100644
index 0000000000..9355fce5e1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/expand_unittest.cc
@@ -0,0 +1,203 @@
+/*
+ *  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.
+ */
+
+// Unit tests for Expand class.
+
+#include "modules/audio_coding/neteq/expand.h"
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+TEST(Expand, CreateAndDestroy) {
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
+}
+
+TEST(Expand, CreateUsingFactory) {
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  ExpandFactory expand_factory;
+  Expand* expand = expand_factory.Create(&bgn, &sync_buffer, &random_vector,
+                                         &statistics, fs, channels);
+  EXPECT_TRUE(expand != NULL);
+  delete expand;
+}
+
+namespace {
+class FakeStatisticsCalculator : public StatisticsCalculator {
+ public:
+  void LogDelayedPacketOutageEvent(int num_samples, int fs_hz) override {
+    last_outage_duration_samples_ = num_samples;
+  }
+
+  int last_outage_duration_samples() const {
+    return last_outage_duration_samples_;
+  }
+
+ private:
+  int last_outage_duration_samples_ = 0;
+};
+
+// This is the same size that is given to the SyncBuffer object in NetEq.
+const size_t kNetEqSyncBufferLengthMs = 720;
+}  // namespace
+
+class ExpandTest : public ::testing::Test {
+ protected:
+  ExpandTest()
+      : input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+                    32000),
+        test_sample_rate_hz_(32000),
+        num_channels_(1),
+        background_noise_(num_channels_),
+        sync_buffer_(num_channels_,
+                     kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
+        expand_(&background_noise_,
+                &sync_buffer_,
+                &random_vector_,
+                &statistics_,
+                test_sample_rate_hz_,
+                num_channels_) {
+    input_file_.set_output_rate_hz(test_sample_rate_hz_);
+  }
+
+  void SetUp() override {
+    // Fast-forward the input file until there is speech (about 1.1 second into
+    // the file).
+    const int speech_start_samples =
+        static_cast<int>(test_sample_rate_hz_ * 1.1f);
+    ASSERT_TRUE(input_file_.Seek(speech_start_samples));
+
+    // Pre-load the sync buffer with speech data.
+    std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
+    ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
+    sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
+    ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
+  }
+
+  test::ResampleInputAudioFile input_file_;
+  int test_sample_rate_hz_;
+  size_t num_channels_;
+  BackgroundNoise background_noise_;
+  SyncBuffer sync_buffer_;
+  RandomVector random_vector_;
+  FakeStatisticsCalculator statistics_;
+  Expand expand_;
+};
+
+// This test calls the expand object to produce concealment data a few times,
+// and then ends by calling SetParametersForNormalAfterExpand. This simulates
+// the situation where the packet next up for decoding was just delayed, not
+// lost.
+TEST_F(ExpandTest, DelayedPacketOutage) {
+  AudioMultiVector output(num_channels_);
+  size_t sum_output_len_samples = 0;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    sum_output_len_samples += output.Size();
+    EXPECT_EQ(0, statistics_.last_outage_duration_samples());
+  }
+  expand_.SetParametersForNormalAfterExpand();
+  // Convert `sum_output_len_samples` to milliseconds.
+  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples),
+            statistics_.last_outage_duration_samples());
+}
+
+// This test is similar to DelayedPacketOutage, but ends by calling
+// SetParametersForMergeAfterExpand. This simulates the situation where the
+// packet next up for decoding was actually lost (or at least a later packet
+// arrived before it).
+TEST_F(ExpandTest, LostPacketOutage) {
+  AudioMultiVector output(num_channels_);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    EXPECT_EQ(0, statistics_.last_outage_duration_samples());
+  }
+  expand_.SetParametersForMergeAfterExpand();
+  EXPECT_EQ(0, statistics_.last_outage_duration_samples());
+}
+
+// This test is similar to the DelayedPacketOutage test above, but with the
+// difference that Expand::Reset() is called after 5 calls to Expand::Process().
+// This should reset the statistics, and will in the end lead to an outage of
+// 5 periods instead of 10.
+TEST_F(ExpandTest, CheckOutageStatsAfterReset) {
+  AudioMultiVector output(num_channels_);
+  size_t sum_output_len_samples = 0;
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(0, expand_.Process(&output));
+    EXPECT_GT(output.Size(), 0u);
+    sum_output_len_samples += output.Size();
+    if (i == 5) {
+      expand_.Reset();
+      sum_output_len_samples = 0;
+    }
+    EXPECT_EQ(0, statistics_.last_outage_duration_samples());
+  }
+  expand_.SetParametersForNormalAfterExpand();
+  // Convert `sum_output_len_samples` to milliseconds.
+  EXPECT_EQ(rtc::checked_cast<int>(sum_output_len_samples),
+            statistics_.last_outage_duration_samples());
+}
+
+namespace {
+// Runs expand until Muted() returns true. Times out after 1000 calls.
+void ExpandUntilMuted(size_t num_channels, Expand* expand) {
+  EXPECT_FALSE(expand->Muted()) << "Instance is muted from the start";
+  AudioMultiVector output(num_channels);
+  int num_calls = 0;
+  while (!expand->Muted()) {
+    ASSERT_LT(num_calls++, 1000) << "Test timed out";
+    EXPECT_EQ(0, expand->Process(&output));
+  }
+}
+}  // namespace
+
+// Verifies that Muted() returns true after a long expand period. Also verifies
+// that Muted() is reset to false after calling Reset(),
+// SetParametersForMergeAfterExpand() and SetParametersForNormalAfterExpand().
+TEST_F(ExpandTest, Muted) {
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.Reset();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.SetParametersForMergeAfterExpand();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+
+  expand_.Reset();  // Must reset in order to start a new expand period.
+  ExpandUntilMuted(num_channels_, &expand_);
+  expand_.SetParametersForNormalAfterExpand();
+  EXPECT_FALSE(expand_.Muted());  // Should be back to unmuted.
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/g3doc/index.md b/third_party/libwebrtc/modules/audio_coding/neteq/g3doc/index.md
new file mode 100644
index 0000000000..40e76e2742
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/g3doc/index.md
@@ -0,0 +1,102 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'jakobi' reviewed: '2021-04-13'} *-->
+
+# NetEq
+
+NetEq is the audio jitter buffer and packet loss concealer. The jitter buffer is
+an adaptive jitter buffer, meaning that the buffering delay is continuously
+optimized based on the network conditions. Its main goal is to ensure a smooth
+playout of incoming audio packets from the network with a low amount of audio
+artifacts (alterations to the original content of the packets) while at the same
+time keep the delay as low as possible.
+
+## API
+
+At a high level, the NetEq API has two main functions:
+[`InsertPacket`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=198;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72)
+and
+[`GetAudio`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=219;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72).
+
+### InsertPacket
+
+[`InsertPacket`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=198;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72)
+delivers an RTP packet from the network to NetEq where the following happens:
+
+1.  The packet is discarded if it is too late for playout (for example if it was
+    reordered). Otherwize it is put into the packet buffer where it is stored
+    until it is time for playout. If the buffer is full, discard all the
+    existing packets (this should be rare).
+2.  The interarrival time between packets is analyzed and statistics is updated
+    which is used to derive a new target playout delay. The interarrival time is
+    measured in the number of GetAudio ‘ticks’ and thus clock drift between the
+    sender and receiver can be accounted for.
+
+### GetAudio
+
+[`GetAudio`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=219;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72)
+pulls 10 ms of audio from NetEq for playout. A much simplified decision logic is
+as follows:
+
+1.  If there is 10 ms audio in the sync buffer then return that.
+2.  If the next packet is available (based on RTP timestamp) in the packet
+    buffer then decode it and append the result to the sync buffer.
+    1.  Compare the current delay estimate (filtered buffer level) with the
+        target delay and time stretch (accelerate or decelerate) the contents of
+        the sync buffer if the buffer level is too high or too low.
+    2.  Return 10 ms of audio from the sync buffer.
+3.  If the last decoded packet was a discontinuous transmission (DTX) packet
+    then generate comfort noise.
+4.  If there is no available packet for decoding due to the next packet having
+    not arrived or been lost then generate packet loss concealment by
+    extrapolating the remaining audio in the sync buffer or by asking the
+    decoder to produce it.
+
+In summary, the output is the result one of the following operations:
+
+*   Normal: audio decoded from a packet.
+*   Acceleration: accelerated playout of a decoded packet.
+*   Preemptive expand: decelerated playout of a decoded packet.
+*   Expand: packet loss concealment generated by NetEq or the decoder.
+*   Merge: audio stitched together from packet loss concealment to decoded data
+    in case of a loss.
+*   Comfort noise (CNG): comfort noise generated by NetEq or the decoder between
+    talk spurts due to discontinuous transmission of packets (DTX).
+
+## Statistics
+
+There are a number of functions that can be used to query the internal state of
+NetEq, statistics about the type of audio output and latency metrics such as how
+long time packets have waited in the buffer.
+
+*   [`NetworkStatistics`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=273;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72):
+    instantaneous values or stats averaged over the duration since last call to
+    this function.
+*   [`GetLifetimeStatistics`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=280;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72):
+    cumulative stats that persist over the lifetime of the class.
+*   [`GetOperationsAndState`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/neteq/neteq.h;l=284;drc=4461f059d180fe8c2886d422ebd1cb55b5c83e72):
+    information about the internal state of NetEq (is only inteded to be used
+    for testing and debugging).
+
+## Tests and tools
+
+*   [`neteq_rtpplay`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc;drc=cee751abff598fc19506f77de08bea7c61b9dcca):
+    Simulate NetEq behavior based on either an RTP dump, a PCAP file or an RTC
+    event log. A replacement audio file can also be used instead of the original
+    payload. Outputs aggregated statistics and optionally an audio file to
+    listen to.
+*   [`neteq_speed_test`](https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc;drc=2ab97f6f8e27b47c0d9beeb8b6ca5387bda9f55c):
+    Measure performance of NetEq, used on perf bots.
+*   Unit tests including bit exactness tests where RTP file is used as an input
+    to NetEq, the output is concatenated and a checksum is calculated and
+    compared against a reference.
+
+## Other responsibilities
+
+*   Dual-tone multi-frequency signaling (DTMF): receive telephone events and
+    produce dual tone waveforms.
+*   Forward error correction (RED or codec inband FEC): split inserted packets
+    and prioritize the payloads.
+*   NACK (negative acknowledgement): keep track of lost packets and generate a
+    list of packets to NACK.
+*   Audio/video sync: NetEq can be instructed to increase the latency in order
+    to keep audio and video in sync.
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/histogram.cc b/third_party/libwebrtc/modules/audio_coding/neteq/histogram.cc
new file mode 100644
index 0000000000..e4b7f10379
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/histogram.cc
@@ -0,0 +1,149 @@
+/*
+ *  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/neteq/histogram.h"
+
+#include <algorithm>
+#include <cstdlib>
+#include <numeric>
+
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+Histogram::Histogram(size_t num_buckets,
+                     int forget_factor,
+                     absl::optional<double> start_forget_weight)
+    : buckets_(num_buckets, 0),
+      forget_factor_(0),
+      base_forget_factor_(forget_factor),
+      add_count_(0),
+      start_forget_weight_(start_forget_weight) {
+  RTC_DCHECK_LT(base_forget_factor_, 1 << 15);
+}
+
+Histogram::~Histogram() {}
+
+// Each element in the vector is first multiplied by the forgetting factor
+// `forget_factor_`. Then the vector element indicated by `iat_packets` is then
+// increased (additive) by 1 - `forget_factor_`. This way, the probability of
+// `value` is slightly increased, while the sum of the histogram remains
+// constant (=1).
+// Due to inaccuracies in the fixed-point arithmetic, the histogram may no
+// longer sum up to 1 (in Q30) after the update. To correct this, a correction
+// term is added or subtracted from the first element (or elements) of the
+// vector.
+// The forgetting factor `forget_factor_` is also updated. When the DelayManager
+// is reset, the factor is set to 0 to facilitate rapid convergence in the
+// beginning. With each update of the histogram, the factor is increased towards
+// the steady-state value `base_forget_factor_`.
+void Histogram::Add(int value) {
+  RTC_DCHECK(value >= 0);
+  RTC_DCHECK(value < static_cast<int>(buckets_.size()));
+  int vector_sum = 0;  // Sum up the vector elements as they are processed.
+  // Multiply each element in `buckets_` with `forget_factor_`.
+  for (int& bucket : buckets_) {
+    bucket = (static_cast<int64_t>(bucket) * forget_factor_) >> 15;
+    vector_sum += bucket;
+  }
+
+  // Increase the probability for the currently observed inter-arrival time
+  // by 1 - `forget_factor_`. The factor is in Q15, `buckets_` in Q30.
+  // Thus, left-shift 15 steps to obtain result in Q30.
+  buckets_[value] += (32768 - forget_factor_) << 15;
+  vector_sum += (32768 - forget_factor_) << 15;  // Add to vector sum.
+
+  // `buckets_` should sum up to 1 (in Q30), but it may not due to
+  // fixed-point rounding errors.
+  vector_sum -= 1 << 30;  // Should be zero. Compensate if not.
+  if (vector_sum != 0) {
+    // Modify a few values early in `buckets_`.
+    int flip_sign = vector_sum > 0 ? -1 : 1;
+    for (int& bucket : buckets_) {
+      // Add/subtract 1/16 of the element, but not more than `vector_sum`.
+      int correction = flip_sign * std::min(std::abs(vector_sum), bucket >> 4);
+      bucket += correction;
+      vector_sum += correction;
+      if (std::abs(vector_sum) == 0) {
+        break;
+      }
+    }
+  }
+  RTC_DCHECK(vector_sum == 0);  // Verify that the above is correct.
+
+  ++add_count_;
+
+  // Update `forget_factor_` (changes only during the first seconds after a
+  // reset). The factor converges to `base_forget_factor_`.
+  if (start_forget_weight_) {
+    if (forget_factor_ != base_forget_factor_) {
+      int old_forget_factor = forget_factor_;
+      int forget_factor =
+          (1 << 15) * (1 - start_forget_weight_.value() / (add_count_ + 1));
+      forget_factor_ =
+          std::max(0, std::min(base_forget_factor_, forget_factor));
+      // The histogram is updated recursively by forgetting the old histogram
+      // with `forget_factor_` and adding a new sample multiplied by |1 -
+      // forget_factor_|. We need to make sure that the effective weight on the
+      // new sample is no smaller than those on the old samples, i.e., to
+      // satisfy the following DCHECK.
+      RTC_DCHECK_GE((1 << 15) - forget_factor_,
+                    ((1 << 15) - old_forget_factor) * forget_factor_ >> 15);
+    }
+  } else {
+    forget_factor_ += (base_forget_factor_ - forget_factor_ + 3) >> 2;
+  }
+}
+
+int Histogram::Quantile(int probability) {
+  // Find the bucket for which the probability of observing an
+  // inter-arrival time larger than or equal to `index` is larger than or
+  // equal to `probability`. The sought probability is estimated using
+  // the histogram as the reverse cumulant PDF, i.e., the sum of elements from
+  // the end up until `index`. Now, since the sum of all elements is 1
+  // (in Q30) by definition, and since the solution is often a low value for
+  // `iat_index`, it is more efficient to start with `sum` = 1 and subtract
+  // elements from the start of the histogram.
+  int inverse_probability = (1 << 30) - probability;
+  size_t index = 0;        // Start from the beginning of `buckets_`.
+  int sum = 1 << 30;       // Assign to 1 in Q30.
+  sum -= buckets_[index];
+
+  while ((sum > inverse_probability) && (index < buckets_.size() - 1)) {
+    // Subtract the probabilities one by one until the sum is no longer greater
+    // than `inverse_probability`.
+    ++index;
+    sum -= buckets_[index];
+  }
+  return static_cast<int>(index);
+}
+
+// Set the histogram vector to an exponentially decaying distribution
+// buckets_[i] = 0.5^(i+1), i = 0, 1, 2, ...
+// buckets_ is in Q30.
+void Histogram::Reset() {
+  // Set temp_prob to (slightly more than) 1 in Q14. This ensures that the sum
+  // of buckets_ is 1.
+  uint16_t temp_prob = 0x4002;  // 16384 + 2 = 100000000000010 binary.
+  for (int& bucket : buckets_) {
+    temp_prob >>= 1;
+    bucket = temp_prob << 16;
+  }
+  forget_factor_ = 0;  // Adapt the histogram faster for the first few packets.
+  add_count_ = 0;
+}
+
+int Histogram::NumBuckets() const {
+  return buckets_.size();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/histogram.h b/third_party/libwebrtc/modules/audio_coding/neteq/histogram.h
new file mode 100644
index 0000000000..265a10e00a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/histogram.h
@@ -0,0 +1,64 @@
+/*
+ *  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_NETEQ_HISTOGRAM_H_
+#define MODULES_AUDIO_CODING_NETEQ_HISTOGRAM_H_
+
+#include <string.h>  // Provide access to size_t.
+
+#include <vector>
+
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+class Histogram {
+ public:
+  // Creates histogram with capacity `num_buckets` and `forget_factor` in Q15.
+  Histogram(size_t num_buckets,
+            int forget_factor,
+            absl::optional<double> start_forget_weight = absl::nullopt);
+
+  virtual ~Histogram();
+
+  // Resets the histogram to the default start distribution.
+  virtual void Reset();
+
+  // Add entry in bucket `index`.
+  virtual void Add(int index);
+
+  // Calculates the quantile at `probability` (in Q30) of the histogram
+  // distribution.
+  virtual int Quantile(int probability);
+
+  // Returns the number of buckets in the histogram.
+  virtual int NumBuckets() const;
+
+  // Returns the probability for each bucket in Q30.
+  const std::vector<int>& buckets() const { return buckets_; }
+
+  // Accessors only intended for testing purposes.
+  int base_forget_factor_for_testing() const { return base_forget_factor_; }
+  int forget_factor_for_testing() const { return forget_factor_; }
+  absl::optional<double> start_forget_weight_for_testing() const {
+    return start_forget_weight_;
+  }
+
+ private:
+  std::vector<int> buckets_;
+  int forget_factor_;  // Q15
+  const int base_forget_factor_;
+  int add_count_;
+  const absl::optional<double> start_forget_weight_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_HISTOGRAM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/histogram_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/histogram_unittest.cc
new file mode 100644
index 0000000000..e30a2956dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/histogram_unittest.cc
@@ -0,0 +1,73 @@
+/*
+ *  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/neteq/histogram.h"
+
+#include <cmath>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(HistogramTest, Initialization) {
+  Histogram histogram(65, 32440);
+  histogram.Reset();
+  const auto& buckets = histogram.buckets();
+  double sum = 0.0;
+  for (size_t i = 0; i < buckets.size(); i++) {
+    EXPECT_NEAR(ldexp(std::pow(0.5, static_cast<int>(i + 1)), 30), buckets[i],
+                65537);
+    // Tolerance 65537 in Q30 corresponds to a delta of approximately 0.00006.
+    sum += buckets[i];
+  }
+  EXPECT_EQ(1 << 30, static_cast<int>(sum));  // Should be 1 in Q30.
+}
+
+TEST(HistogramTest, Add) {
+  Histogram histogram(10, 32440);
+  histogram.Reset();
+  const std::vector<int> before = histogram.buckets();
+  const int index = 5;
+  histogram.Add(index);
+  const std::vector<int> after = histogram.buckets();
+  EXPECT_GT(after[index], before[index]);
+  int sum = 0;
+  for (int bucket : after) {
+    sum += bucket;
+  }
+  EXPECT_EQ(1 << 30, sum);
+}
+
+TEST(HistogramTest, ForgetFactor) {
+  Histogram histogram(10, 32440);
+  histogram.Reset();
+  const std::vector<int> before = histogram.buckets();
+  const int index = 4;
+  histogram.Add(index);
+  const std::vector<int> after = histogram.buckets();
+  for (int i = 0; i < histogram.NumBuckets(); ++i) {
+    if (i != index) {
+      EXPECT_LT(after[i], before[i]);
+    }
+  }
+}
+
+TEST(HistogramTest, ReachSteadyStateForgetFactor) {
+  static constexpr int kSteadyStateForgetFactor = (1 << 15) * 0.9993;
+  Histogram histogram(100, kSteadyStateForgetFactor, 1.0);
+  histogram.Reset();
+  int n = (1 << 15) / ((1 << 15) - kSteadyStateForgetFactor);
+  for (int i = 0; i < n; ++i) {
+    histogram.Add(0);
+  }
+  EXPECT_EQ(histogram.forget_factor_for_testing(), kSteadyStateForgetFactor);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/merge.cc b/third_party/libwebrtc/modules/audio_coding/neteq/merge.cc
new file mode 100644
index 0000000000..0aec6d2597
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/merge.cc
@@ -0,0 +1,391 @@
+/*
+ *  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/neteq/merge.h"
+
+#include <string.h>  // memmove, memcpy, memset, size_t
+
+#include <algorithm>  // min, max
+#include <memory>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+Merge::Merge(int fs_hz,
+             size_t num_channels,
+             Expand* expand,
+             SyncBuffer* sync_buffer)
+    : fs_hz_(fs_hz),
+      num_channels_(num_channels),
+      fs_mult_(fs_hz_ / 8000),
+      timestamps_per_call_(static_cast<size_t>(fs_hz_ / 100)),
+      expand_(expand),
+      sync_buffer_(sync_buffer),
+      expanded_(num_channels_) {
+  RTC_DCHECK_GT(num_channels_, 0);
+}
+
+Merge::~Merge() = default;
+
+size_t Merge::Process(int16_t* input,
+                      size_t input_length,
+                      AudioMultiVector* output) {
+  // TODO(hlundin): Change to an enumerator and skip assert.
+  RTC_DCHECK(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
+             fs_hz_ == 48000);
+  RTC_DCHECK_LE(fs_hz_, kMaxSampleRate);  // Should not be possible.
+  if (input_length == 0) {
+    return 0;
+  }
+
+  size_t old_length;
+  size_t expand_period;
+  // Get expansion data to overlap and mix with.
+  size_t expanded_length = GetExpandedSignal(&old_length, &expand_period);
+
+  // Transfer input signal to an AudioMultiVector.
+  AudioMultiVector input_vector(num_channels_);
+  input_vector.PushBackInterleaved(
+      rtc::ArrayView<const int16_t>(input, input_length));
+  size_t input_length_per_channel = input_vector.Size();
+  RTC_DCHECK_EQ(input_length_per_channel, input_length / num_channels_);
+
+  size_t best_correlation_index = 0;
+  size_t output_length = 0;
+
+  std::unique_ptr<int16_t[]> input_channel(
+      new int16_t[input_length_per_channel]);
+  std::unique_ptr<int16_t[]> expanded_channel(new int16_t[expanded_length]);
+  for (size_t channel = 0; channel < num_channels_; ++channel) {
+    input_vector[channel].CopyTo(input_length_per_channel, 0,
+                                 input_channel.get());
+    expanded_[channel].CopyTo(expanded_length, 0, expanded_channel.get());
+
+    const int16_t new_mute_factor = std::min<int16_t>(
+        16384, SignalScaling(input_channel.get(), input_length_per_channel,
+                             expanded_channel.get()));
+
+    if (channel == 0) {
+      // Downsample, correlate, and find strongest correlation period for the
+      // reference (i.e., first) channel only.
+      // Downsample to 4kHz sample rate.
+      Downsample(input_channel.get(), input_length_per_channel,
+                 expanded_channel.get(), expanded_length);
+
+      // Calculate the lag of the strongest correlation period.
+      best_correlation_index = CorrelateAndPeakSearch(
+          old_length, input_length_per_channel, expand_period);
+    }
+
+    temp_data_.resize(input_length_per_channel + best_correlation_index);
+    int16_t* decoded_output = temp_data_.data() + best_correlation_index;
+
+    // Mute the new decoded data if needed (and unmute it linearly).
+    // This is the overlapping part of expanded_signal.
+    size_t interpolation_length =
+        std::min(kMaxCorrelationLength * fs_mult_,
+                 expanded_length - best_correlation_index);
+    interpolation_length =
+        std::min(interpolation_length, input_length_per_channel);
+
+    RTC_DCHECK_LE(new_mute_factor, 16384);
+    int16_t mute_factor =
+        std::max(expand_->MuteFactor(channel), new_mute_factor);
+    RTC_DCHECK_GE(mute_factor, 0);
+
+    if (mute_factor < 16384) {
+      // Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
+      // and so on, or as fast as it takes to come back to full gain within the
+      // frame length.
+      const int back_to_fullscale_inc = static_cast<int>(
+          ((16384 - mute_factor) << 6) / input_length_per_channel);
+      const int increment = std::max(4194 / fs_mult_, back_to_fullscale_inc);
+      mute_factor = static_cast<int16_t>(DspHelper::RampSignal(
+          input_channel.get(), interpolation_length, mute_factor, increment));
+      DspHelper::UnmuteSignal(&input_channel[interpolation_length],
+                              input_length_per_channel - interpolation_length,
+                              &mute_factor, increment,
+                              &decoded_output[interpolation_length]);
+    } else {
+      // No muting needed.
+      memmove(
+          &decoded_output[interpolation_length],
+          &input_channel[interpolation_length],
+          sizeof(int16_t) * (input_length_per_channel - interpolation_length));
+    }
+
+    // Do overlap and mix linearly.
+    int16_t increment =
+        static_cast<int16_t>(16384 / (interpolation_length + 1));  // In Q14.
+    int16_t local_mute_factor = 16384 - increment;
+    memmove(temp_data_.data(), expanded_channel.get(),
+            sizeof(int16_t) * best_correlation_index);
+    DspHelper::CrossFade(&expanded_channel[best_correlation_index],
+                         input_channel.get(), interpolation_length,
+                         &local_mute_factor, increment, decoded_output);
+
+    output_length = best_correlation_index + input_length_per_channel;
+    if (channel == 0) {
+      RTC_DCHECK(output->Empty());  // Output should be empty at this point.
+      output->AssertSize(output_length);
+    } else {
+      RTC_DCHECK_EQ(output->Size(), output_length);
+    }
+    (*output)[channel].OverwriteAt(temp_data_.data(), output_length, 0);
+  }
+
+  // Copy back the first part of the data to `sync_buffer_` and remove it from
+  // `output`.
+  sync_buffer_->ReplaceAtIndex(*output, old_length, sync_buffer_->next_index());
+  output->PopFront(old_length);
+
+  // Return new added length. `old_length` samples were borrowed from
+  // `sync_buffer_`.
+  RTC_DCHECK_GE(output_length, old_length);
+  return output_length - old_length;
+}
+
+size_t Merge::GetExpandedSignal(size_t* old_length, size_t* expand_period) {
+  // Check how much data that is left since earlier.
+  *old_length = sync_buffer_->FutureLength();
+  // Should never be less than overlap_length.
+  RTC_DCHECK_GE(*old_length, expand_->overlap_length());
+  // Generate data to merge the overlap with using expand.
+  expand_->SetParametersForMergeAfterExpand();
+
+  if (*old_length >= 210 * kMaxSampleRate / 8000) {
+    // TODO(hlundin): Write test case for this.
+    // The number of samples available in the sync buffer is more than what fits
+    // in expanded_signal. Keep the first 210 * kMaxSampleRate / 8000 samples,
+    // but shift them towards the end of the buffer. This is ok, since all of
+    // the buffer will be expand data anyway, so as long as the beginning is
+    // left untouched, we're fine.
+    size_t length_diff = *old_length - 210 * kMaxSampleRate / 8000;
+    sync_buffer_->InsertZerosAtIndex(length_diff, sync_buffer_->next_index());
+    *old_length = 210 * kMaxSampleRate / 8000;
+    // This is the truncated length.
+  }
+  // This assert should always be true thanks to the if statement above.
+  RTC_DCHECK_GE(210 * kMaxSampleRate / 8000, *old_length);
+
+  AudioMultiVector expanded_temp(num_channels_);
+  expand_->Process(&expanded_temp);
+  *expand_period = expanded_temp.Size();  // Samples per channel.
+
+  expanded_.Clear();
+  // Copy what is left since earlier into the expanded vector.
+  expanded_.PushBackFromIndex(*sync_buffer_, sync_buffer_->next_index());
+  RTC_DCHECK_EQ(expanded_.Size(), *old_length);
+  RTC_DCHECK_GT(expanded_temp.Size(), 0);
+  // Do "ugly" copy and paste from the expanded in order to generate more data
+  // to correlate (but not interpolate) with.
+  const size_t required_length = static_cast<size_t>((120 + 80 + 2) * fs_mult_);
+  if (expanded_.Size() < required_length) {
+    while (expanded_.Size() < required_length) {
+      // Append one more pitch period each time.
+      expanded_.PushBack(expanded_temp);
+    }
+    // Trim the length to exactly `required_length`.
+    expanded_.PopBack(expanded_.Size() - required_length);
+  }
+  RTC_DCHECK_GE(expanded_.Size(), required_length);
+  return required_length;
+}
+
+int16_t Merge::SignalScaling(const int16_t* input,
+                             size_t input_length,
+                             const int16_t* expanded_signal) const {
+  // Adjust muting factor if new vector is more or less of the BGN energy.
+  const auto mod_input_length = rtc::SafeMin<size_t>(
+      64 * rtc::dchecked_cast<size_t>(fs_mult_), input_length);
+
+  // Missing input, do no muting
+  if (mod_input_length == 0) {
+    return 16384;
+  }
+
+  const int16_t expanded_max =
+      WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
+  int32_t factor =
+      (expanded_max * expanded_max) / (std::numeric_limits<int32_t>::max() /
+                                       static_cast<int32_t>(mod_input_length));
+  const int expanded_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+  int32_t energy_expanded = WebRtcSpl_DotProductWithScale(
+      expanded_signal, expanded_signal, mod_input_length, expanded_shift);
+
+  // Calculate energy of input signal.
+  const int16_t input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
+  factor = (input_max * input_max) / (std::numeric_limits<int32_t>::max() /
+                                      static_cast<int32_t>(mod_input_length));
+  const int input_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
+  int32_t energy_input = WebRtcSpl_DotProductWithScale(
+      input, input, mod_input_length, input_shift);
+
+  // Align to the same Q-domain.
+  if (input_shift > expanded_shift) {
+    energy_expanded = energy_expanded >> (input_shift - expanded_shift);
+  } else {
+    energy_input = energy_input >> (expanded_shift - input_shift);
+  }
+
+  // Calculate muting factor to use for new frame.
+  int16_t mute_factor;
+  if (energy_input > energy_expanded) {
+    // Normalize `energy_input` to 14 bits.
+    int16_t temp_shift = WebRtcSpl_NormW32(energy_input) - 17;
+    energy_input = WEBRTC_SPL_SHIFT_W32(energy_input, temp_shift);
+    // Put `energy_expanded` in a domain 14 higher, so that
+    // energy_expanded / energy_input is in Q14.
+    energy_expanded = WEBRTC_SPL_SHIFT_W32(energy_expanded, temp_shift + 14);
+    // Calculate sqrt(energy_expanded / energy_input) in Q14.
+    mute_factor = static_cast<int16_t>(
+        WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14));
+  } else {
+    // Set to 1 (in Q14) when `expanded` has higher energy than `input`.
+    mute_factor = 16384;
+  }
+
+  return mute_factor;
+}
+
+// TODO(hlundin): There are some parameter values in this method that seem
+// strange. Compare with Expand::Correlation.
+void Merge::Downsample(const int16_t* input,
+                       size_t input_length,
+                       const int16_t* expanded_signal,
+                       size_t expanded_length) {
+  const int16_t* filter_coefficients;
+  size_t num_coefficients;
+  int decimation_factor = fs_hz_ / 4000;
+  static const size_t kCompensateDelay = 0;
+  size_t length_limit = static_cast<size_t>(fs_hz_ / 100);  // 10 ms in samples.
+  if (fs_hz_ == 8000) {
+    filter_coefficients = DspHelper::kDownsample8kHzTbl;
+    num_coefficients = 3;
+  } else if (fs_hz_ == 16000) {
+    filter_coefficients = DspHelper::kDownsample16kHzTbl;
+    num_coefficients = 5;
+  } else if (fs_hz_ == 32000) {
+    filter_coefficients = DspHelper::kDownsample32kHzTbl;
+    num_coefficients = 7;
+  } else {  // fs_hz_ == 48000
+    filter_coefficients = DspHelper::kDownsample48kHzTbl;
+    num_coefficients = 7;
+  }
+  size_t signal_offset = num_coefficients - 1;
+  WebRtcSpl_DownsampleFast(
+      &expanded_signal[signal_offset], expanded_length - signal_offset,
+      expanded_downsampled_, kExpandDownsampLength, filter_coefficients,
+      num_coefficients, decimation_factor, kCompensateDelay);
+  if (input_length <= length_limit) {
+    // Not quite long enough, so we have to cheat a bit.
+    // If the input is shorter than the offset, we consider the input to be 0
+    // length. This will cause us to skip the downsampling since it makes no
+    // sense anyway, and input_downsampled_ will be filled with zeros. This is
+    // clearly a pathological case, and the signal quality will suffer, but
+    // there is not much we can do.
+    const size_t temp_len =
+        input_length > signal_offset ? input_length - signal_offset : 0;
+    // TODO(hlundin): Should `downsamp_temp_len` be corrected for round-off
+    // errors? I.e., (temp_len + decimation_factor - 1) / decimation_factor?
+    size_t downsamp_temp_len = temp_len / decimation_factor;
+    if (downsamp_temp_len > 0) {
+      WebRtcSpl_DownsampleFast(&input[signal_offset], temp_len,
+                               input_downsampled_, downsamp_temp_len,
+                               filter_coefficients, num_coefficients,
+                               decimation_factor, kCompensateDelay);
+    }
+    memset(&input_downsampled_[downsamp_temp_len], 0,
+           sizeof(int16_t) * (kInputDownsampLength - downsamp_temp_len));
+  } else {
+    WebRtcSpl_DownsampleFast(
+        &input[signal_offset], input_length - signal_offset, input_downsampled_,
+        kInputDownsampLength, filter_coefficients, num_coefficients,
+        decimation_factor, kCompensateDelay);
+  }
+}
+
+size_t Merge::CorrelateAndPeakSearch(size_t start_position,
+                                     size_t input_length,
+                                     size_t expand_period) const {
+  // Calculate correlation without any normalization.
+  const size_t max_corr_length = kMaxCorrelationLength;
+  size_t stop_position_downsamp =
+      std::min(max_corr_length, expand_->max_lag() / (fs_mult_ * 2) + 1);
+
+  int32_t correlation[kMaxCorrelationLength];
+  CrossCorrelationWithAutoShift(input_downsampled_, expanded_downsampled_,
+                                kInputDownsampLength, stop_position_downsamp, 1,
+                                correlation);
+
+  // Normalize correlation to 14 bits and copy to a 16-bit array.
+  const size_t pad_length = expand_->overlap_length() - 1;
+  const size_t correlation_buffer_size = 2 * pad_length + kMaxCorrelationLength;
+  std::unique_ptr<int16_t[]> correlation16(
+      new int16_t[correlation_buffer_size]);
+  memset(correlation16.get(), 0, correlation_buffer_size * sizeof(int16_t));
+  int16_t* correlation_ptr = &correlation16[pad_length];
+  int32_t max_correlation =
+      WebRtcSpl_MaxAbsValueW32(correlation, stop_position_downsamp);
+  int norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
+  WebRtcSpl_VectorBitShiftW32ToW16(correlation_ptr, stop_position_downsamp,
+                                   correlation, norm_shift);
+
+  // Calculate allowed starting point for peak finding.
+  // The peak location bestIndex must fulfill two criteria:
+  // (1) w16_bestIndex + input_length <
+  //     timestamps_per_call_ + expand_->overlap_length();
+  // (2) w16_bestIndex + input_length < start_position.
+  size_t start_index = timestamps_per_call_ + expand_->overlap_length();
+  start_index = std::max(start_position, start_index);
+  start_index = (input_length > start_index) ? 0 : (start_index - input_length);
+  // Downscale starting index to 4kHz domain. (fs_mult_ * 2 = fs_hz_ / 4000.)
+  size_t start_index_downsamp = start_index / (fs_mult_ * 2);
+
+  // Calculate a modified `stop_position_downsamp` to account for the increased
+  // start index `start_index_downsamp` and the effective array length.
+  size_t modified_stop_pos =
+      std::min(stop_position_downsamp,
+               kMaxCorrelationLength + pad_length - start_index_downsamp);
+  size_t best_correlation_index;
+  int16_t best_correlation;
+  static const size_t kNumCorrelationCandidates = 1;
+  DspHelper::PeakDetection(&correlation_ptr[start_index_downsamp],
+                           modified_stop_pos, kNumCorrelationCandidates,
+                           fs_mult_, &best_correlation_index,
+                           &best_correlation);
+  // Compensate for modified start index.
+  best_correlation_index += start_index;
+
+  // Ensure that underrun does not occur for 10ms case => we have to get at
+  // least 10ms + overlap . (This should never happen thanks to the above
+  // modification of peak-finding starting point.)
+  while (((best_correlation_index + input_length) <
+          (timestamps_per_call_ + expand_->overlap_length())) ||
+         ((best_correlation_index + input_length) < start_position)) {
+    RTC_DCHECK_NOTREACHED();                  // Should never happen.
+    best_correlation_index += expand_period;  // Jump one lag ahead.
+  }
+  return best_correlation_index;
+}
+
+size_t Merge::RequiredFutureSamples() {
+  return fs_hz_ / 100 * num_channels_;  // 10 ms.
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/merge.h b/third_party/libwebrtc/modules/audio_coding/neteq/merge.h
new file mode 100644
index 0000000000..2f27106bfe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/merge.h
@@ -0,0 +1,101 @@
+/*
+ *  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_NETEQ_MERGE_H_
+#define MODULES_AUDIO_CODING_NETEQ_MERGE_H_
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class Expand;
+class SyncBuffer;
+
+// This class handles the transition from expansion to normal operation.
+// When a packet is not available for decoding when needed, the expand operation
+// is called to generate extrapolation data. If the missing packet arrives,
+// i.e., it was just delayed, it can be decoded and appended directly to the
+// end of the expanded data (thanks to how the Expand class operates). However,
+// if a later packet arrives instead, the loss is a fact, and the new data must
+// be stitched together with the end of the expanded data. This stitching is
+// what the Merge class does.
+class Merge {
+ public:
+  Merge(int fs_hz,
+        size_t num_channels,
+        Expand* expand,
+        SyncBuffer* sync_buffer);
+  virtual ~Merge();
+
+  Merge(const Merge&) = delete;
+  Merge& operator=(const Merge&) = delete;
+
+  // The main method to produce the audio data. The decoded data is supplied in
+  // `input`, having `input_length` samples in total for all channels
+  // (interleaved). The result is written to `output`. The number of channels
+  // allocated in `output` defines the number of channels that will be used when
+  // de-interleaving `input`.
+  virtual size_t Process(int16_t* input,
+                         size_t input_length,
+                         AudioMultiVector* output);
+
+  virtual size_t RequiredFutureSamples();
+
+ protected:
+  const int fs_hz_;
+  const size_t num_channels_;
+
+ private:
+  static const int kMaxSampleRate = 48000;
+  static const size_t kExpandDownsampLength = 100;
+  static const size_t kInputDownsampLength = 40;
+  static const size_t kMaxCorrelationLength = 60;
+
+  // Calls `expand_` to get more expansion data to merge with. The data is
+  // written to `expanded_signal_`. Returns the length of the expanded data,
+  // while `expand_period` will be the number of samples in one expansion period
+  // (typically one pitch period). The value of `old_length` will be the number
+  // of samples that were taken from the `sync_buffer_`.
+  size_t GetExpandedSignal(size_t* old_length, size_t* expand_period);
+
+  // Analyzes `input` and `expanded_signal` and returns muting factor (Q14) to
+  // be used on the new data.
+  int16_t SignalScaling(const int16_t* input,
+                        size_t input_length,
+                        const int16_t* expanded_signal) const;
+
+  // Downsamples `input` (`input_length` samples) and `expanded_signal` to
+  // 4 kHz sample rate. The downsampled signals are written to
+  // `input_downsampled_` and `expanded_downsampled_`, respectively.
+  void Downsample(const int16_t* input,
+                  size_t input_length,
+                  const int16_t* expanded_signal,
+                  size_t expanded_length);
+
+  // Calculates cross-correlation between `input_downsampled_` and
+  // `expanded_downsampled_`, and finds the correlation maximum. The maximizing
+  // lag is returned.
+  size_t CorrelateAndPeakSearch(size_t start_position,
+                                size_t input_length,
+                                size_t expand_period) const;
+
+  const int fs_mult_;  // fs_hz_ / 8000.
+  const size_t timestamps_per_call_;
+  Expand* expand_;
+  SyncBuffer* sync_buffer_;
+  int16_t expanded_downsampled_[kExpandDownsampLength];
+  int16_t input_downsampled_[kInputDownsampLength];
+  AudioMultiVector expanded_;
+  std::vector<int16_t> temp_data_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MERGE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/merge_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/merge_unittest.cc
new file mode 100644
index 0000000000..d5a55eb056
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/merge_unittest.cc
@@ -0,0 +1,121 @@
+/*
+ *  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.
+ */
+
+// Unit tests for Merge class.
+
+#include "modules/audio_coding/neteq/merge.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+TEST(Merge, CreateAndDestroy) {
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
+  Merge merge(fs, channels, &expand, &sync_buffer);
+}
+
+namespace {
+// This is the same size that is given to the SyncBuffer object in NetEq.
+const size_t kNetEqSyncBufferLengthMs = 720;
+}  // namespace
+
+class MergeTest : public testing::TestWithParam<size_t> {
+ protected:
+  MergeTest()
+      : input_file_(test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+                    32000),
+        test_sample_rate_hz_(8000),
+        num_channels_(1),
+        background_noise_(num_channels_),
+        sync_buffer_(num_channels_,
+                     kNetEqSyncBufferLengthMs * test_sample_rate_hz_ / 1000),
+        expand_(&background_noise_,
+                &sync_buffer_,
+                &random_vector_,
+                &statistics_,
+                test_sample_rate_hz_,
+                num_channels_),
+        merge_(test_sample_rate_hz_, num_channels_, &expand_, &sync_buffer_) {
+    input_file_.set_output_rate_hz(test_sample_rate_hz_);
+  }
+
+  void SetUp() override {
+    // Fast-forward the input file until there is speech (about 1.1 second into
+    // the file).
+    const int speech_start_samples =
+        static_cast<int>(test_sample_rate_hz_ * 1.1f);
+    ASSERT_TRUE(input_file_.Seek(speech_start_samples));
+
+    // Pre-load the sync buffer with speech data.
+    std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]);
+    ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get()));
+    sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0);
+    // Move index such that the sync buffer appears to have 5 ms left to play.
+    sync_buffer_.set_next_index(sync_buffer_.next_index() -
+                                test_sample_rate_hz_ * 5 / 1000);
+    ASSERT_EQ(1u, num_channels_) << "Fix: Must populate all channels.";
+    ASSERT_GT(sync_buffer_.FutureLength(), 0u);
+  }
+
+  test::ResampleInputAudioFile input_file_;
+  int test_sample_rate_hz_;
+  size_t num_channels_;
+  BackgroundNoise background_noise_;
+  SyncBuffer sync_buffer_;
+  RandomVector random_vector_;
+  StatisticsCalculator statistics_;
+  Expand expand_;
+  Merge merge_;
+};
+
+TEST_P(MergeTest, Process) {
+  AudioMultiVector output(num_channels_);
+  // Start by calling Expand once, to prime the state.
+  EXPECT_EQ(0, expand_.Process(&output));
+  EXPECT_GT(output.Size(), 0u);
+  output.Clear();
+  // Now call Merge, but with a very short decoded input. Try different length
+  // if the input.
+  const size_t input_len = GetParam();
+  std::vector<int16_t> input(input_len, 17);
+  merge_.Process(input.data(), input_len, &output);
+  EXPECT_GT(output.Size(), 0u);
+}
+
+// Instantiate with values for the input length that are interesting in
+// Merge::Downsample. Why are these values interesting?
+// - In 8000 Hz sample rate, signal_offset in Merge::Downsample will be 2, so
+//   the values 1, 2, 3 are just around that value.
+// - Also in 8000 Hz, the variable length_limit in the same method will be 80,
+//   so values 80 and 81 will be on either side of the branch point
+//   "input_length <= length_limit".
+// - Finally, 160 is simply 20 ms in 8000 Hz, which is a common packet size.
+INSTANTIATE_TEST_SUITE_P(DifferentInputLengths,
+                         MergeTest,
+                         testing::Values(1, 2, 3, 80, 81, 160));
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
new file mode 100644
index 0000000000..503f6ac6bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockBufferLevelFilter : public BufferLevelFilter {
+ public:
+  MOCK_METHOD(void,
+              Update,
+              (size_t buffer_size_samples, int time_stretched_samples));
+  MOCK_METHOD(int, filtered_current_level, (), (const));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
new file mode 100644
index 0000000000..2394120e99
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
@@ -0,0 +1,50 @@
+/*
+ *  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_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
+
+#include <string>
+
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDecoderDatabase : public DecoderDatabase {
+ public:
+  explicit MockDecoderDatabase(
+      rtc::scoped_refptr<AudioDecoderFactory> factory = nullptr)
+      : DecoderDatabase(factory, absl::nullopt) {}
+  ~MockDecoderDatabase() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(bool, Empty, (), (const, override));
+  MOCK_METHOD(int, Size, (), (const, override));
+  MOCK_METHOD(int,
+              RegisterPayload,
+              (int rtp_payload_type, const SdpAudioFormat& audio_format),
+              (override));
+  MOCK_METHOD(int, Remove, (uint8_t rtp_payload_type), (override));
+  MOCK_METHOD(void, RemoveAll, (), (override));
+  MOCK_METHOD(const DecoderInfo*,
+              GetDecoderInfo,
+              (uint8_t rtp_payload_type),
+              (const, override));
+  MOCK_METHOD(int,
+              SetActiveDecoder,
+              (uint8_t rtp_payload_type, bool* new_decoder),
+              (override));
+  MOCK_METHOD(AudioDecoder*, GetActiveDecoder, (), (const, override));
+  MOCK_METHOD(int, SetActiveCngDecoder, (uint8_t rtp_payload_type), (override));
+  MOCK_METHOD(ComfortNoiseDecoder*, GetActiveCngDecoder, (), (const, override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
new file mode 100644
index 0000000000..d783f8743b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
+
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDelayManager : public DelayManager {
+ public:
+  MockDelayManager(const MockDelayManager::Config& config,
+                   const TickTimer* tick_timer)
+      : DelayManager(config, tick_timer) {}
+  MOCK_METHOD(int, TargetDelayMs, (), (const));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
new file mode 100644
index 0000000000..c60c56d36b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
+
+#include "modules/audio_coding/neteq/dtmf_buffer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDtmfBuffer : public DtmfBuffer {
+ public:
+  MockDtmfBuffer(int fs) : DtmfBuffer(fs) {}
+  ~MockDtmfBuffer() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(void, Flush, (), (override));
+  MOCK_METHOD(int, InsertEvent, (const DtmfEvent& event), (override));
+  MOCK_METHOD(bool,
+              GetEvent,
+              (uint32_t current_timestamp, DtmfEvent* event),
+              (override));
+  MOCK_METHOD(size_t, Length, (), (const, override));
+  MOCK_METHOD(bool, Empty, (), (const, override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
new file mode 100644
index 0000000000..60de167c29
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
@@ -0,0 +1,33 @@
+/*
+ *  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_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
+
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockDtmfToneGenerator : public DtmfToneGenerator {
+ public:
+  ~MockDtmfToneGenerator() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(int, Init, (int fs, int event, int attenuation), (override));
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(int,
+              Generate,
+              (size_t num_samples, AudioMultiVector* output),
+              (override));
+  MOCK_METHOD(bool, initialized, (), (const, override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_expand.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_expand.h
new file mode 100644
index 0000000000..9d66779021
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_expand.h
@@ -0,0 +1,60 @@
+/*
+ *  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_NETEQ_MOCK_MOCK_EXPAND_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
+
+#include "modules/audio_coding/neteq/expand.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockExpand : public Expand {
+ public:
+  MockExpand(BackgroundNoise* background_noise,
+             SyncBuffer* sync_buffer,
+             RandomVector* random_vector,
+             StatisticsCalculator* statistics,
+             int fs,
+             size_t num_channels)
+      : Expand(background_noise,
+               sync_buffer,
+               random_vector,
+               statistics,
+               fs,
+               num_channels) {}
+  ~MockExpand() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(int, Process, (AudioMultiVector * output), (override));
+  MOCK_METHOD(void, SetParametersForNormalAfterExpand, (), (override));
+  MOCK_METHOD(void, SetParametersForMergeAfterExpand, (), (override));
+  MOCK_METHOD(size_t, overlap_length, (), (const, override));
+};
+
+}  // namespace webrtc
+
+namespace webrtc {
+
+class MockExpandFactory : public ExpandFactory {
+ public:
+  MOCK_METHOD(Expand*,
+              Create,
+              (BackgroundNoise * background_noise,
+               SyncBuffer* sync_buffer,
+               RandomVector* random_vector,
+               StatisticsCalculator* statistics,
+               int fs,
+               size_t num_channels),
+              (const, override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_histogram.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_histogram.h
new file mode 100644
index 0000000000..03abbc1d4b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_histogram.h
@@ -0,0 +1,30 @@
+/*
+ *  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_NETEQ_MOCK_MOCK_HISTOGRAM_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_HISTOGRAM_H_
+
+#include "modules/audio_coding/neteq/histogram.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockHistogram : public Histogram {
+ public:
+  MockHistogram(size_t num_buckets, int forget_factor)
+      : Histogram(num_buckets, forget_factor) {}
+  virtual ~MockHistogram() {}
+
+  MOCK_METHOD(void, Add, (int), (override));
+  MOCK_METHOD(int, Quantile, (int), (override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_HISTOGRAM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_neteq_controller.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_neteq_controller.h
new file mode 100644
index 0000000000..6d88e09216
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_neteq_controller.h
@@ -0,0 +1,62 @@
+/*
+ *  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_NETEQ_MOCK_MOCK_NETEQ_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_NETEQ_CONTROLLER_H_
+
+#include "api/neteq/neteq_controller.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockNetEqController : public NetEqController {
+ public:
+  MockNetEqController() = default;
+  ~MockNetEqController() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(void, SoftReset, (), (override));
+  MOCK_METHOD(NetEq::Operation,
+              GetDecision,
+              (const NetEqStatus& neteq_status, bool* reset_decoder),
+              (override));
+  MOCK_METHOD(void, RegisterEmptyPacket, (), (override));
+  MOCK_METHOD(void,
+              SetSampleRate,
+              (int fs_hz, size_t output_size_samples),
+              (override));
+  MOCK_METHOD(bool, SetMaximumDelay, (int delay_ms), (override));
+  MOCK_METHOD(bool, SetMinimumDelay, (int delay_ms), (override));
+  MOCK_METHOD(bool, SetBaseMinimumDelay, (int delay_ms), (override));
+  MOCK_METHOD(int, GetBaseMinimumDelay, (), (const, override));
+  MOCK_METHOD(bool, CngRfc3389On, (), (const, override));
+  MOCK_METHOD(bool, CngOff, (), (const, override));
+  MOCK_METHOD(void, SetCngOff, (), (override));
+  MOCK_METHOD(void, ExpandDecision, (NetEq::Operation operation), (override));
+  MOCK_METHOD(void, AddSampleMemory, (int32_t value), (override));
+  MOCK_METHOD(int, TargetLevelMs, (), (const, override));
+  MOCK_METHOD(absl::optional<int>,
+              PacketArrived,
+              (int fs_hz,
+               bool should_update_stats,
+               const PacketArrivedInfo& info),
+              (override));
+  MOCK_METHOD(void, NotifyMutedState, (), (override));
+  MOCK_METHOD(bool, PeakFound, (), (const, override));
+  MOCK_METHOD(int, GetFilteredBufferLevel, (), (const, override));
+  MOCK_METHOD(void, set_sample_memory, (int32_t value), (override));
+  MOCK_METHOD(size_t, noise_fast_forward, (), (const, override));
+  MOCK_METHOD(size_t, packet_length_samples, (), (const, override));
+  MOCK_METHOD(void, set_packet_length_samples, (size_t value), (override));
+  MOCK_METHOD(void, set_prev_time_scale, (bool value), (override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_NETEQ_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
new file mode 100644
index 0000000000..48357ea466
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
@@ -0,0 +1,82 @@
+/*
+ *  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_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockPacketBuffer : public PacketBuffer {
+ public:
+  MockPacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer)
+      : PacketBuffer(max_number_of_packets, tick_timer) {}
+  ~MockPacketBuffer() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(void, Flush, (StatisticsCalculator * stats), (override));
+  MOCK_METHOD(void,
+              PartialFlush,
+              (int target_level_ms,
+               size_t sample_rate,
+               size_t last_decoded_length,
+               StatisticsCalculator* stats),
+              (override));
+  MOCK_METHOD(bool, Empty, (), (const, override));
+  MOCK_METHOD(int,
+              InsertPacket,
+              (Packet && packet,
+               StatisticsCalculator* stats,
+               size_t last_decoded_length,
+               size_t sample_rate,
+               int target_level_ms,
+               const DecoderDatabase& decoder_database),
+              (override));
+  MOCK_METHOD(int,
+              InsertPacketList,
+              (PacketList * packet_list,
+               const DecoderDatabase& decoder_database,
+               absl::optional<uint8_t>* current_rtp_payload_type,
+               absl::optional<uint8_t>* current_cng_rtp_payload_type,
+               StatisticsCalculator* stats,
+               size_t last_decoded_length,
+               size_t sample_rate,
+               int target_level_ms),
+              (override));
+  MOCK_METHOD(int,
+              NextTimestamp,
+              (uint32_t * next_timestamp),
+              (const, override));
+  MOCK_METHOD(int,
+              NextHigherTimestamp,
+              (uint32_t timestamp, uint32_t* next_timestamp),
+              (const, override));
+  MOCK_METHOD(const Packet*, PeekNextPacket, (), (const, override));
+  MOCK_METHOD(absl::optional<Packet>, GetNextPacket, (), (override));
+  MOCK_METHOD(int,
+              DiscardNextPacket,
+              (StatisticsCalculator * stats),
+              (override));
+  MOCK_METHOD(void,
+              DiscardOldPackets,
+              (uint32_t timestamp_limit,
+               uint32_t horizon_samples,
+               StatisticsCalculator* stats),
+              (override));
+  MOCK_METHOD(void,
+              DiscardAllOldPackets,
+              (uint32_t timestamp_limit, StatisticsCalculator* stats),
+              (override));
+  MOCK_METHOD(size_t, NumPacketsInBuffer, (), (const, override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
new file mode 100644
index 0000000000..9daf571a80
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
@@ -0,0 +1,30 @@
+/*
+ *  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_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
+
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockRedPayloadSplitter : public RedPayloadSplitter {
+ public:
+  MOCK_METHOD(bool, SplitRed, (PacketList * packet_list), (override));
+  MOCK_METHOD(void,
+              CheckRedPayloads,
+              (PacketList * packet_list,
+               const DecoderDatabase& decoder_database),
+              (override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_statistics_calculator.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
new file mode 100644
index 0000000000..f8812478d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
@@ -0,0 +1,30 @@
+/*
+ *  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_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
+
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockStatisticsCalculator : public StatisticsCalculator {
+ public:
+  MOCK_METHOD(void, PacketsDiscarded, (size_t num_packets), (override));
+  MOCK_METHOD(void,
+              SecondaryPacketsDiscarded,
+              (size_t num_packets),
+              (override));
+  MOCK_METHOD(void, RelativePacketArrivalDelay, (size_t delay_ms), (override));
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.cc b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.cc
new file mode 100644
index 0000000000..04cc5b52e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.cc
@@ -0,0 +1,267 @@
+/*
+ *  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 "modules/audio_coding/neteq/nack_tracker.h"
+
+#include <cstdint>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+const int kDefaultSampleRateKhz = 48;
+const int kMaxPacketSizeMs = 120;
+constexpr char kNackTrackerConfigFieldTrial[] =
+    "WebRTC-Audio-NetEqNackTrackerConfig";
+
+}  // namespace
+
+NackTracker::Config::Config() {
+  auto parser = StructParametersParser::Create(
+      "packet_loss_forget_factor", &packet_loss_forget_factor,
+      "ms_per_loss_percent", &ms_per_loss_percent, "never_nack_multiple_times",
+      &never_nack_multiple_times, "require_valid_rtt", &require_valid_rtt,
+      "max_loss_rate", &max_loss_rate);
+  parser->Parse(
+      webrtc::field_trial::FindFullName(kNackTrackerConfigFieldTrial));
+  RTC_LOG(LS_INFO) << "Nack tracker config:"
+                      " packet_loss_forget_factor="
+                   << packet_loss_forget_factor
+                   << " ms_per_loss_percent=" << ms_per_loss_percent
+                   << " never_nack_multiple_times=" << never_nack_multiple_times
+                   << " require_valid_rtt=" << require_valid_rtt
+                   << " max_loss_rate=" << max_loss_rate;
+}
+
+NackTracker::NackTracker()
+    : sequence_num_last_received_rtp_(0),
+      timestamp_last_received_rtp_(0),
+      any_rtp_received_(false),
+      sequence_num_last_decoded_rtp_(0),
+      timestamp_last_decoded_rtp_(0),
+      any_rtp_decoded_(false),
+      sample_rate_khz_(kDefaultSampleRateKhz),
+      max_nack_list_size_(kNackListSizeLimit) {}
+
+NackTracker::~NackTracker() = default;
+
+void NackTracker::UpdateSampleRate(int sample_rate_hz) {
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  sample_rate_khz_ = sample_rate_hz / 1000;
+}
+
+void NackTracker::UpdateLastReceivedPacket(uint16_t sequence_number,
+                                           uint32_t timestamp) {
+  // Just record the value of sequence number and timestamp if this is the
+  // first packet.
+  if (!any_rtp_received_) {
+    sequence_num_last_received_rtp_ = sequence_number;
+    timestamp_last_received_rtp_ = timestamp;
+    any_rtp_received_ = true;
+    // If no packet is decoded, to have a reasonable estimate of time-to-play
+    // use the given values.
+    if (!any_rtp_decoded_) {
+      sequence_num_last_decoded_rtp_ = sequence_number;
+      timestamp_last_decoded_rtp_ = timestamp;
+    }
+    return;
+  }
+
+  if (sequence_number == sequence_num_last_received_rtp_)
+    return;
+
+  // Received RTP should not be in the list.
+  nack_list_.erase(sequence_number);
+
+  // If this is an old sequence number, no more action is required, return.
+  if (IsNewerSequenceNumber(sequence_num_last_received_rtp_, sequence_number))
+    return;
+
+  UpdatePacketLossRate(sequence_number - sequence_num_last_received_rtp_ - 1);
+
+  UpdateList(sequence_number, timestamp);
+
+  sequence_num_last_received_rtp_ = sequence_number;
+  timestamp_last_received_rtp_ = timestamp;
+  LimitNackListSize();
+}
+
+absl::optional<int> NackTracker::GetSamplesPerPacket(
+    uint16_t sequence_number_current_received_rtp,
+    uint32_t timestamp_current_received_rtp) const {
+  uint32_t timestamp_increase =
+      timestamp_current_received_rtp - timestamp_last_received_rtp_;
+  uint16_t sequence_num_increase =
+      sequence_number_current_received_rtp - sequence_num_last_received_rtp_;
+
+  int samples_per_packet = timestamp_increase / sequence_num_increase;
+  if (samples_per_packet == 0 ||
+      samples_per_packet > kMaxPacketSizeMs * sample_rate_khz_) {
+    // Not a valid samples per packet.
+    return absl::nullopt;
+  }
+  return samples_per_packet;
+}
+
+void NackTracker::UpdateList(uint16_t sequence_number_current_received_rtp,
+                             uint32_t timestamp_current_received_rtp) {
+  if (!IsNewerSequenceNumber(sequence_number_current_received_rtp,
+                             sequence_num_last_received_rtp_ + 1)) {
+    return;
+  }
+  RTC_DCHECK(!any_rtp_decoded_ ||
+             IsNewerSequenceNumber(sequence_number_current_received_rtp,
+                                   sequence_num_last_decoded_rtp_));
+
+  absl::optional<int> samples_per_packet = GetSamplesPerPacket(
+      sequence_number_current_received_rtp, timestamp_current_received_rtp);
+  if (!samples_per_packet) {
+    return;
+  }
+
+  for (uint16_t n = sequence_num_last_received_rtp_ + 1;
+       IsNewerSequenceNumber(sequence_number_current_received_rtp, n); ++n) {
+    uint32_t timestamp = EstimateTimestamp(n, *samples_per_packet);
+    NackElement nack_element(TimeToPlay(timestamp), timestamp);
+    nack_list_.insert(nack_list_.end(), std::make_pair(n, nack_element));
+  }
+}
+
+uint32_t NackTracker::EstimateTimestamp(uint16_t sequence_num,
+                                        int samples_per_packet) {
+  uint16_t sequence_num_diff = sequence_num - sequence_num_last_received_rtp_;
+  return sequence_num_diff * samples_per_packet + timestamp_last_received_rtp_;
+}
+
+void NackTracker::UpdateEstimatedPlayoutTimeBy10ms() {
+  while (!nack_list_.empty() &&
+         nack_list_.begin()->second.time_to_play_ms <= 10)
+    nack_list_.erase(nack_list_.begin());
+
+  for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end(); ++it)
+    it->second.time_to_play_ms -= 10;
+}
+
+void NackTracker::UpdateLastDecodedPacket(uint16_t sequence_number,
+                                          uint32_t timestamp) {
+  if (IsNewerSequenceNumber(sequence_number, sequence_num_last_decoded_rtp_) ||
+      !any_rtp_decoded_) {
+    sequence_num_last_decoded_rtp_ = sequence_number;
+    timestamp_last_decoded_rtp_ = timestamp;
+    // Packets in the list with sequence numbers less than the
+    // sequence number of the decoded RTP should be removed from the lists.
+    // They will be discarded by the jitter buffer if they arrive.
+    nack_list_.erase(nack_list_.begin(),
+                     nack_list_.upper_bound(sequence_num_last_decoded_rtp_));
+
+    // Update estimated time-to-play.
+    for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end();
+         ++it)
+      it->second.time_to_play_ms = TimeToPlay(it->second.estimated_timestamp);
+  } else {
+    RTC_DCHECK_EQ(sequence_number, sequence_num_last_decoded_rtp_);
+
+    // Same sequence number as before. 10 ms is elapsed, update estimations for
+    // time-to-play.
+    UpdateEstimatedPlayoutTimeBy10ms();
+
+    // Update timestamp for better estimate of time-to-play, for packets which
+    // are added to NACK list later on.
+    timestamp_last_decoded_rtp_ += sample_rate_khz_ * 10;
+  }
+  any_rtp_decoded_ = true;
+}
+
+NackTracker::NackList NackTracker::GetNackList() const {
+  return nack_list_;
+}
+
+void NackTracker::Reset() {
+  nack_list_.clear();
+
+  sequence_num_last_received_rtp_ = 0;
+  timestamp_last_received_rtp_ = 0;
+  any_rtp_received_ = false;
+  sequence_num_last_decoded_rtp_ = 0;
+  timestamp_last_decoded_rtp_ = 0;
+  any_rtp_decoded_ = false;
+  sample_rate_khz_ = kDefaultSampleRateKhz;
+}
+
+void NackTracker::SetMaxNackListSize(size_t max_nack_list_size) {
+  RTC_CHECK_GT(max_nack_list_size, 0);
+  // Ugly hack to get around the problem of passing static consts by reference.
+  const size_t kNackListSizeLimitLocal = NackTracker::kNackListSizeLimit;
+  RTC_CHECK_LE(max_nack_list_size, kNackListSizeLimitLocal);
+
+  max_nack_list_size_ = max_nack_list_size;
+  LimitNackListSize();
+}
+
+void NackTracker::LimitNackListSize() {
+  uint16_t limit = sequence_num_last_received_rtp_ -
+                   static_cast<uint16_t>(max_nack_list_size_) - 1;
+  nack_list_.erase(nack_list_.begin(), nack_list_.upper_bound(limit));
+}
+
+int64_t NackTracker::TimeToPlay(uint32_t timestamp) const {
+  uint32_t timestamp_increase = timestamp - timestamp_last_decoded_rtp_;
+  return timestamp_increase / sample_rate_khz_;
+}
+
+// We don't erase elements with time-to-play shorter than round-trip-time.
+std::vector<uint16_t> NackTracker::GetNackList(int64_t round_trip_time_ms) {
+  RTC_DCHECK_GE(round_trip_time_ms, 0);
+  std::vector<uint16_t> sequence_numbers;
+  if (round_trip_time_ms == 0) {
+    if (config_.require_valid_rtt) {
+      return sequence_numbers;
+    } else {
+      round_trip_time_ms = config_.default_rtt_ms;
+    }
+  }
+  if (packet_loss_rate_ >
+      static_cast<uint32_t>(config_.max_loss_rate * (1 << 30))) {
+    return sequence_numbers;
+  }
+  // The estimated packet loss is between 0 and 1, so we need to multiply by 100
+  // here.
+  int max_wait_ms =
+      100.0 * config_.ms_per_loss_percent * packet_loss_rate_ / (1 << 30);
+  for (NackList::const_iterator it = nack_list_.begin(); it != nack_list_.end();
+       ++it) {
+    int64_t time_since_packet_ms =
+        (timestamp_last_received_rtp_ - it->second.estimated_timestamp) /
+        sample_rate_khz_;
+    if (it->second.time_to_play_ms > round_trip_time_ms ||
+        time_since_packet_ms + round_trip_time_ms < max_wait_ms)
+      sequence_numbers.push_back(it->first);
+  }
+  if (config_.never_nack_multiple_times) {
+    nack_list_.clear();
+  }
+  return sequence_numbers;
+}
+
+void NackTracker::UpdatePacketLossRate(int packets_lost) {
+  const uint64_t alpha_q30 = (1 << 30) * config_.packet_loss_forget_factor;
+  // Exponential filter.
+  packet_loss_rate_ = (alpha_q30 * packet_loss_rate_) >> 30;
+  for (int i = 0; i < packets_lost; ++i) {
+    packet_loss_rate_ =
+        ((alpha_q30 * packet_loss_rate_) >> 30) + ((1 << 30) - alpha_q30);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.h b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.h
new file mode 100644
index 0000000000..14ba2166d1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker.h
@@ -0,0 +1,211 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
+#define MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <map>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/include/module_common_types_public.h"
+#include "rtc_base/gtest_prod_util.h"
+
+//
+// The NackTracker class keeps track of the lost packets, an estimate of
+// time-to-play for each packet is also given.
+//
+// Every time a packet is pushed into NetEq, LastReceivedPacket() has to be
+// called to update the NACK list.
+//
+// Every time 10ms audio is pulled from NetEq LastDecodedPacket() should be
+// called, and time-to-play is updated at that moment.
+//
+// If packet N is received, any packet prior to N which has not arrived is
+// considered lost, and should be labeled as "missing" (the size of
+// the list might be limited and older packet eliminated from the list).
+//
+// The NackTracker class has to know about the sample rate of the packets to
+// compute time-to-play. So sample rate should be set as soon as the first
+// packet is received. If there is a change in the receive codec (sender changes
+// codec) then NackTracker should be reset. This is because NetEQ would flush
+// its buffer and re-transmission is meaning less for old packet. Therefore, in
+// that case, after reset the sampling rate has to be updated.
+//
+// Thread Safety
+// =============
+// Please note that this class in not thread safe. The class must be protected
+// if different APIs are called from different threads.
+//
+namespace webrtc {
+
+class NackTracker {
+ public:
+  // A limit for the size of the NACK list.
+  static const size_t kNackListSizeLimit = 500;  // 10 seconds for 20 ms frame
+                                                 // packets.
+  NackTracker();
+  ~NackTracker();
+
+  // Set a maximum for the size of the NACK list. If the last received packet
+  // has sequence number of N, then NACK list will not contain any element
+  // with sequence number earlier than N - `max_nack_list_size`.
+  //
+  // The largest maximum size is defined by `kNackListSizeLimit`
+  void SetMaxNackListSize(size_t max_nack_list_size);
+
+  // Set the sampling rate.
+  //
+  // If associated sampling rate of the received packets is changed, call this
+  // function to update sampling rate. Note that if there is any change in
+  // received codec then NetEq will flush its buffer and NACK has to be reset.
+  // After Reset() is called sampling rate has to be set.
+  void UpdateSampleRate(int sample_rate_hz);
+
+  // Update the sequence number and the timestamp of the last decoded RTP. This
+  // API should be called every time 10 ms audio is pulled from NetEq.
+  void UpdateLastDecodedPacket(uint16_t sequence_number, uint32_t timestamp);
+
+  // Update the sequence number and the timestamp of the last received RTP. This
+  // API should be called every time a packet pushed into ACM.
+  void UpdateLastReceivedPacket(uint16_t sequence_number, uint32_t timestamp);
+
+  // Get a list of "missing" packets which have expected time-to-play larger
+  // than the given round-trip-time (in milliseconds).
+  // Note: Late packets are not included.
+  // Calling this method multiple times may give different results, since the
+  // internal nack list may get flushed if never_nack_multiple_times_ is true.
+  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms);
+
+  // Reset to default values. The NACK list is cleared.
+  // `max_nack_list_size_` preserves its value.
+  void Reset();
+
+  // Returns the estimated packet loss rate in Q30, for testing only.
+  uint32_t GetPacketLossRateForTest() { return packet_loss_rate_; }
+
+ private:
+  // This test need to access the private method GetNackList().
+  FRIEND_TEST_ALL_PREFIXES(NackTrackerTest, EstimateTimestampAndTimeToPlay);
+
+  // Options that can be configured via field trial.
+  struct Config {
+    Config();
+
+    // The exponential decay factor used to estimate the packet loss rate.
+    double packet_loss_forget_factor = 0.996;
+    // How many additional ms we are willing to wait (at most) for nacked
+    // packets for each additional percentage of packet loss.
+    int ms_per_loss_percent = 20;
+    // If true, never nack packets more than once.
+    bool never_nack_multiple_times = false;
+    // Only nack if the RTT is valid.
+    bool require_valid_rtt = false;
+    // Default RTT to use unless `require_valid_rtt` is set.
+    int default_rtt_ms = 100;
+    // Do not nack if the loss rate is above this value.
+    double max_loss_rate = 1.0;
+  };
+
+  struct NackElement {
+    NackElement(int64_t initial_time_to_play_ms, uint32_t initial_timestamp)
+        : time_to_play_ms(initial_time_to_play_ms),
+          estimated_timestamp(initial_timestamp) {}
+
+    // Estimated time (ms) left for this packet to be decoded. This estimate is
+    // updated every time jitter buffer decodes a packet.
+    int64_t time_to_play_ms;
+
+    // A guess about the timestamp of the missing packet, it is used for
+    // estimation of `time_to_play_ms`. The estimate might be slightly wrong if
+    // there has been frame-size change since the last received packet and the
+    // missing packet. However, the risk of this is low, and in case of such
+    // errors, there will be a minor misestimation in time-to-play of missing
+    // packets. This will have a very minor effect on NACK performance.
+    uint32_t estimated_timestamp;
+  };
+
+  class NackListCompare {
+   public:
+    bool operator()(uint16_t sequence_number_old,
+                    uint16_t sequence_number_new) const {
+      return IsNewerSequenceNumber(sequence_number_new, sequence_number_old);
+    }
+  };
+
+  typedef std::map<uint16_t, NackElement, NackListCompare> NackList;
+
+  // This API is used only for testing to assess whether time-to-play is
+  // computed correctly.
+  NackList GetNackList() const;
+
+  // This function subtracts 10 ms of time-to-play for all packets in NACK list.
+  // This is called when 10 ms elapsed with no new RTP packet decoded.
+  void UpdateEstimatedPlayoutTimeBy10ms();
+
+  // Returns a valid number of samples per packet given the current received
+  // sequence number and timestamp or nullopt of none could be computed.
+  absl::optional<int> GetSamplesPerPacket(
+      uint16_t sequence_number_current_received_rtp,
+      uint32_t timestamp_current_received_rtp) const;
+
+  // Given the `sequence_number_current_received_rtp` of currently received RTP
+  // update the list. Packets that are older than the received packet are added
+  // to the nack list.
+  void UpdateList(uint16_t sequence_number_current_received_rtp,
+                  uint32_t timestamp_current_received_rtp);
+
+  // Packets which have sequence number older that
+  // `sequence_num_last_received_rtp_` - `max_nack_list_size_` are removed
+  // from the NACK list.
+  void LimitNackListSize();
+
+  // Estimate timestamp of a missing packet given its sequence number.
+  uint32_t EstimateTimestamp(uint16_t sequence_number, int samples_per_packet);
+
+  // Compute time-to-play given a timestamp.
+  int64_t TimeToPlay(uint32_t timestamp) const;
+
+  // Updates the estimated packet lost rate.
+  void UpdatePacketLossRate(int packets_lost);
+
+  const Config config_;
+
+  // Valid if a packet is received.
+  uint16_t sequence_num_last_received_rtp_;
+  uint32_t timestamp_last_received_rtp_;
+  bool any_rtp_received_;  // If any packet received.
+
+  // Valid if a packet is decoded.
+  uint16_t sequence_num_last_decoded_rtp_;
+  uint32_t timestamp_last_decoded_rtp_;
+  bool any_rtp_decoded_;  // If any packet decoded.
+
+  int sample_rate_khz_;  // Sample rate in kHz.
+
+  // A list of missing packets to be retransmitted. Components of the list
+  // contain the sequence number of missing packets and the estimated time that
+  // each pack is going to be played out.
+  NackList nack_list_;
+
+  // NACK list will not keep track of missing packets prior to
+  // `sequence_num_last_received_rtp_` - `max_nack_list_size_`.
+  size_t max_nack_list_size_;
+
+  // Current estimate of the packet loss rate in Q30.
+  uint32_t packet_loss_rate_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc
new file mode 100644
index 0000000000..bcc5120ff3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc
@@ -0,0 +1,565 @@
+/*
+ *  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 "modules/audio_coding/neteq/nack_tracker.h"
+
+#include <stdint.h>
+
+#include <algorithm>
+#include <memory>
+
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+const int kSampleRateHz = 16000;
+const int kPacketSizeMs = 30;
+const uint32_t kTimestampIncrement = 480;  // 30 ms.
+const int64_t kShortRoundTripTimeMs = 1;
+
+bool IsNackListCorrect(const std::vector<uint16_t>& nack_list,
+                       const uint16_t* lost_sequence_numbers,
+                       size_t num_lost_packets) {
+  if (nack_list.size() != num_lost_packets)
+    return false;
+
+  if (num_lost_packets == 0)
+    return true;
+
+  for (size_t k = 0; k < nack_list.size(); ++k) {
+    int seq_num = nack_list[k];
+    bool seq_num_matched = false;
+    for (size_t n = 0; n < num_lost_packets; ++n) {
+      if (seq_num == lost_sequence_numbers[n]) {
+        seq_num_matched = true;
+        break;
+      }
+    }
+    if (!seq_num_matched)
+      return false;
+  }
+  return true;
+}
+
+}  // namespace
+
+TEST(NackTrackerTest, EmptyListWhenNoPacketLoss) {
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+
+  int seq_num = 1;
+  uint32_t timestamp = 0;
+
+  std::vector<uint16_t> nack_list;
+  for (int n = 0; n < 100; n++) {
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    seq_num++;
+    timestamp += kTimestampIncrement;
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(nack_list.empty());
+  }
+}
+
+TEST(NackTrackerTest, LatePacketsMovedToNackThenNackListDoesNotChange) {
+  const uint16_t kSequenceNumberLostPackets[] = {2, 3, 4, 5, 6, 7, 8, 9};
+  static const int kNumAllLostPackets = sizeof(kSequenceNumberLostPackets) /
+                                        sizeof(kSequenceNumberLostPackets[0]);
+
+  for (int k = 0; k < 2; k++) {  // Two iteration with/without wrap around.
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+
+    uint16_t sequence_num_lost_packets[kNumAllLostPackets];
+    for (int n = 0; n < kNumAllLostPackets; n++) {
+      sequence_num_lost_packets[n] =
+          kSequenceNumberLostPackets[n] +
+          k * 65531;  // Have wrap around in sequence numbers for |k == 1|.
+    }
+    uint16_t seq_num = sequence_num_lost_packets[0] - 1;
+
+    uint32_t timestamp = 0;
+    std::vector<uint16_t> nack_list;
+
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(nack_list.empty());
+
+    seq_num = sequence_num_lost_packets[kNumAllLostPackets - 1] + 1;
+    timestamp += kTimestampIncrement * (kNumAllLostPackets + 1);
+    int num_lost_packets = std::max(0, kNumAllLostPackets);
+
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(IsNackListCorrect(nack_list, sequence_num_lost_packets,
+                                  num_lost_packets));
+    seq_num++;
+    timestamp += kTimestampIncrement;
+    num_lost_packets++;
+
+    for (int n = 0; n < 100; ++n) {
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      EXPECT_TRUE(IsNackListCorrect(nack_list, sequence_num_lost_packets,
+                                    kNumAllLostPackets));
+      seq_num++;
+      timestamp += kTimestampIncrement;
+    }
+  }
+}
+
+TEST(NackTrackerTest, ArrivedPacketsAreRemovedFromNackList) {
+  const uint16_t kSequenceNumberLostPackets[] = {2, 3, 4, 5, 6, 7, 8, 9};
+  static const int kNumAllLostPackets = sizeof(kSequenceNumberLostPackets) /
+                                        sizeof(kSequenceNumberLostPackets[0]);
+
+  for (int k = 0; k < 2; ++k) {  // Two iteration with/without wrap around.
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+
+    uint16_t sequence_num_lost_packets[kNumAllLostPackets];
+    for (int n = 0; n < kNumAllLostPackets; ++n) {
+      sequence_num_lost_packets[n] = kSequenceNumberLostPackets[n] +
+                                     k * 65531;  // Wrap around for |k == 1|.
+    }
+
+    uint16_t seq_num = sequence_num_lost_packets[0] - 1;
+    uint32_t timestamp = 0;
+
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(nack_list.empty());
+
+    size_t index_retransmitted_rtp = 0;
+    uint32_t timestamp_retransmitted_rtp = timestamp + kTimestampIncrement;
+
+    seq_num = sequence_num_lost_packets[kNumAllLostPackets - 1] + 1;
+    timestamp += kTimestampIncrement * (kNumAllLostPackets + 1);
+    size_t num_lost_packets = kNumAllLostPackets;
+    for (int n = 0; n < kNumAllLostPackets; ++n) {
+      // Number of lost packets does not change for the first
+      // |kNackThreshold + 1| packets, one is added to the list and one is
+      // removed. Thereafter, the list shrinks every iteration.
+      if (n >= 1)
+        num_lost_packets--;
+
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      EXPECT_TRUE(IsNackListCorrect(
+          nack_list, &sequence_num_lost_packets[index_retransmitted_rtp],
+          num_lost_packets));
+      seq_num++;
+      timestamp += kTimestampIncrement;
+
+      // Retransmission of a lost RTP.
+      nack.UpdateLastReceivedPacket(
+          sequence_num_lost_packets[index_retransmitted_rtp],
+          timestamp_retransmitted_rtp);
+      index_retransmitted_rtp++;
+      timestamp_retransmitted_rtp += kTimestampIncrement;
+
+      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      EXPECT_TRUE(IsNackListCorrect(
+          nack_list, &sequence_num_lost_packets[index_retransmitted_rtp],
+          num_lost_packets - 1));  // One less lost packet in the list.
+    }
+    ASSERT_TRUE(nack_list.empty());
+  }
+}
+
+// Assess if estimation of timestamps and time-to-play is correct. Introduce all
+// combinations that timestamps and sequence numbers might have wrap around.
+TEST(NackTrackerTest, EstimateTimestampAndTimeToPlay) {
+  const uint16_t kLostPackets[] = {2, 3,  4,  5,  6,  7,  8,
+                                   9, 10, 11, 12, 13, 14, 15};
+  static const int kNumAllLostPackets =
+      sizeof(kLostPackets) / sizeof(kLostPackets[0]);
+
+  for (int k = 0; k < 4; ++k) {
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+
+    // Sequence number wrap around if `k` is 2 or 3;
+    int seq_num_offset = (k < 2) ? 0 : 65531;
+
+    // Timestamp wrap around if `k` is 1 or 3.
+    uint32_t timestamp_offset =
+        (k & 0x1) ? static_cast<uint32_t>(0xffffffff) - 6 : 0;
+
+    uint32_t timestamp_lost_packets[kNumAllLostPackets];
+    uint16_t seq_num_lost_packets[kNumAllLostPackets];
+    for (int n = 0; n < kNumAllLostPackets; ++n) {
+      timestamp_lost_packets[n] =
+          timestamp_offset + kLostPackets[n] * kTimestampIncrement;
+      seq_num_lost_packets[n] = seq_num_offset + kLostPackets[n];
+    }
+
+    // We and to push two packets before lost burst starts.
+    uint16_t seq_num = seq_num_lost_packets[0] - 2;
+    uint32_t timestamp = timestamp_lost_packets[0] - 2 * kTimestampIncrement;
+
+    const uint16_t first_seq_num = seq_num;
+    const uint32_t first_timestamp = timestamp;
+
+    // Two consecutive packets to have a correct estimate of timestamp increase.
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    seq_num++;
+    timestamp += kTimestampIncrement;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+    // A packet after the last one which is supposed to be lost.
+    seq_num = seq_num_lost_packets[kNumAllLostPackets - 1] + 1;
+    timestamp =
+        timestamp_lost_packets[kNumAllLostPackets - 1] + kTimestampIncrement;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+    NackTracker::NackList nack_list = nack.GetNackList();
+    EXPECT_EQ(static_cast<size_t>(kNumAllLostPackets), nack_list.size());
+
+    // Pretend the first packet is decoded.
+    nack.UpdateLastDecodedPacket(first_seq_num, first_timestamp);
+    nack_list = nack.GetNackList();
+
+    NackTracker::NackList::iterator it = nack_list.begin();
+    while (it != nack_list.end()) {
+      seq_num = it->first - seq_num_offset;
+      int index = seq_num - kLostPackets[0];
+      EXPECT_EQ(timestamp_lost_packets[index], it->second.estimated_timestamp);
+      EXPECT_EQ((index + 2) * kPacketSizeMs, it->second.time_to_play_ms);
+      ++it;
+    }
+
+    // Pretend 10 ms is passed, and we had pulled audio from NetEq, it still
+    // reports the same sequence number as decoded, time-to-play should be
+    // updated by 10 ms.
+    nack.UpdateLastDecodedPacket(first_seq_num, first_timestamp);
+    nack_list = nack.GetNackList();
+    it = nack_list.begin();
+    while (it != nack_list.end()) {
+      seq_num = it->first - seq_num_offset;
+      int index = seq_num - kLostPackets[0];
+      EXPECT_EQ((index + 2) * kPacketSizeMs - 10, it->second.time_to_play_ms);
+      ++it;
+    }
+  }
+}
+
+TEST(NackTrackerTest,
+     MissingPacketsPriorToLastDecodedRtpShouldNotBeInNackList) {
+  for (int m = 0; m < 2; ++m) {
+    uint16_t seq_num_offset = (m == 0) ? 0 : 65531;  // Wrap around if `m` is 1.
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+
+    // Two consecutive packets to have a correct estimate of timestamp increase.
+    uint16_t seq_num = 0;
+    nack.UpdateLastReceivedPacket(seq_num_offset + seq_num,
+                                  seq_num * kTimestampIncrement);
+    seq_num++;
+    nack.UpdateLastReceivedPacket(seq_num_offset + seq_num,
+                                  seq_num * kTimestampIncrement);
+
+    // Skip 10 packets (larger than NACK threshold).
+    const int kNumLostPackets = 10;
+    seq_num += kNumLostPackets + 1;
+    nack.UpdateLastReceivedPacket(seq_num_offset + seq_num,
+                                  seq_num * kTimestampIncrement);
+
+    const size_t kExpectedListSize = kNumLostPackets;
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_EQ(kExpectedListSize, nack_list.size());
+
+    for (int k = 0; k < 2; ++k) {
+      // Decoding of the first and the second arrived packets.
+      for (int n = 0; n < kPacketSizeMs / 10; ++n) {
+        nack.UpdateLastDecodedPacket(seq_num_offset + k,
+                                     k * kTimestampIncrement);
+        nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+        EXPECT_EQ(kExpectedListSize, nack_list.size());
+      }
+    }
+
+    // Decoding of the last received packet.
+    nack.UpdateLastDecodedPacket(seq_num + seq_num_offset,
+                                 seq_num * kTimestampIncrement);
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(nack_list.empty());
+
+    // Make sure list of late packets is also empty. To check that, push few
+    // packets, if the late list is not empty its content will pop up in NACK
+    // list.
+    for (int n = 0; n < 10; ++n) {
+      seq_num++;
+      nack.UpdateLastReceivedPacket(seq_num_offset + seq_num,
+                                    seq_num * kTimestampIncrement);
+      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      EXPECT_TRUE(nack_list.empty());
+    }
+  }
+}
+
+TEST(NackTrackerTest, Reset) {
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+
+  // Two consecutive packets to have a correct estimate of timestamp increase.
+  uint16_t seq_num = 0;
+  nack.UpdateLastReceivedPacket(seq_num, seq_num * kTimestampIncrement);
+  seq_num++;
+  nack.UpdateLastReceivedPacket(seq_num, seq_num * kTimestampIncrement);
+
+  // Skip 10 packets (larger than NACK threshold).
+  const int kNumLostPackets = 10;
+  seq_num += kNumLostPackets + 1;
+  nack.UpdateLastReceivedPacket(seq_num, seq_num * kTimestampIncrement);
+
+  const size_t kExpectedListSize = kNumLostPackets;
+  std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+  EXPECT_EQ(kExpectedListSize, nack_list.size());
+
+  nack.Reset();
+  nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+  EXPECT_TRUE(nack_list.empty());
+}
+
+TEST(NackTrackerTest, ListSizeAppliedFromBeginning) {
+  const size_t kNackListSize = 10;
+  for (int m = 0; m < 2; ++m) {
+    uint16_t seq_num_offset = (m == 0) ? 0 : 65525;  // Wrap around if `m` is 1.
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+    nack.SetMaxNackListSize(kNackListSize);
+
+    uint16_t seq_num = seq_num_offset;
+    uint32_t timestamp = 0x12345678;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+    // Packet lost more than NACK-list size limit.
+    uint16_t num_lost_packets = kNackListSize + 5;
+
+    seq_num += num_lost_packets + 1;
+    timestamp += (num_lost_packets + 1) * kTimestampIncrement;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_EQ(kNackListSize, nack_list.size());
+  }
+}
+
+TEST(NackTrackerTest, ChangeOfListSizeAppliedAndOldElementsRemoved) {
+  const size_t kNackListSize = 10;
+  for (int m = 0; m < 2; ++m) {
+    uint16_t seq_num_offset = (m == 0) ? 0 : 65525;  // Wrap around if `m` is 1.
+    NackTracker nack;
+    nack.UpdateSampleRate(kSampleRateHz);
+
+    uint16_t seq_num = seq_num_offset;
+    uint32_t timestamp = 0x87654321;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+    // Packet lost more than NACK-list size limit.
+    uint16_t num_lost_packets = kNackListSize + 5;
+
+    std::unique_ptr<uint16_t[]> seq_num_lost(new uint16_t[num_lost_packets]);
+    for (int n = 0; n < num_lost_packets; ++n) {
+      seq_num_lost[n] = ++seq_num;
+    }
+
+    ++seq_num;
+    timestamp += (num_lost_packets + 1) * kTimestampIncrement;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    size_t expected_size = num_lost_packets;
+
+    std::vector<uint16_t> nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_EQ(expected_size, nack_list.size());
+
+    nack.SetMaxNackListSize(kNackListSize);
+    expected_size = kNackListSize;
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(IsNackListCorrect(
+        nack_list, &seq_num_lost[num_lost_packets - kNackListSize],
+        expected_size));
+
+    // NACK list should shrink.
+    for (size_t n = 1; n < kNackListSize; ++n) {
+      ++seq_num;
+      timestamp += kTimestampIncrement;
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+      --expected_size;
+      nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+      EXPECT_TRUE(IsNackListCorrect(
+          nack_list, &seq_num_lost[num_lost_packets - kNackListSize + n],
+          expected_size));
+    }
+
+    // After this packet, NACK list should be empty.
+    ++seq_num;
+    timestamp += kTimestampIncrement;
+    nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    nack_list = nack.GetNackList(kShortRoundTripTimeMs);
+    EXPECT_TRUE(nack_list.empty());
+  }
+}
+
+TEST(NackTrackerTest, RoudTripTimeIsApplied) {
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+  // Packet lost more than NACK-list size limit.
+  uint16_t kNumLostPackets = 5;
+
+  seq_num += (1 + kNumLostPackets);
+  timestamp += (1 + kNumLostPackets) * kTimestampIncrement;
+  nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+  // Expected time-to-play are:
+  // kPacketSizeMs - 10, 2*kPacketSizeMs - 10, 3*kPacketSizeMs - 10, ...
+  //
+  // sequence number:  1,  2,  3,   4,   5
+  // time-to-play:    20, 50, 80, 110, 140
+  //
+  std::vector<uint16_t> nack_list = nack.GetNackList(100);
+  ASSERT_EQ(2u, nack_list.size());
+  EXPECT_EQ(4, nack_list[0]);
+  EXPECT_EQ(5, nack_list[1]);
+}
+
+// Set never_nack_multiple_times to true with a field trial and verify that
+// packets are not nacked multiple times.
+TEST(NackTrackerTest, DoNotNackMultipleTimes) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-NetEqNackTrackerConfig/"
+      "packet_loss_forget_factor:0.996,ms_per_loss_percent:20,"
+      "never_nack_multiple_times:true/");
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+  uint16_t kNumLostPackets = 3;
+
+  seq_num += (1 + kNumLostPackets);
+  timestamp += (1 + kNumLostPackets) * kTimestampIncrement;
+  nack.UpdateLastReceivedPacket(seq_num, timestamp);
+
+  std::vector<uint16_t> nack_list = nack.GetNackList(10);
+  ASSERT_EQ(3u, nack_list.size());
+  EXPECT_EQ(1, nack_list[0]);
+  EXPECT_EQ(2, nack_list[1]);
+  EXPECT_EQ(3, nack_list[2]);
+  // When we get the nack list again, it should be empty.
+  std::vector<uint16_t> nack_list2 = nack.GetNackList(10);
+  EXPECT_TRUE(nack_list2.empty());
+}
+
+// Test if estimated packet loss rate is correct.
+TEST(NackTrackerTest, PacketLossRateCorrect) {
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  auto add_packet = [&nack, &seq_num, &timestamp](bool received) {
+    if (received) {
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    }
+    seq_num++;
+    timestamp += kTimestampIncrement;
+  };
+  // Add some packets, but every fourth packet is lost.
+  for (int i = 0; i < 300; i++) {
+    add_packet(true);
+    add_packet(true);
+    add_packet(true);
+    add_packet(false);
+  }
+  // 1 << 28 is 0.25 in Q30. We expect the packet loss estimate to be within
+  // 0.01 of that.
+  EXPECT_NEAR(nack.GetPacketLossRateForTest(), 1 << 28, (1 << 30) / 100);
+}
+
+TEST(NackTrackerTest, DoNotNackAfterDtx) {
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  nack.UpdateLastReceivedPacket(seq_num, timestamp);
+  EXPECT_TRUE(nack.GetNackList(0).empty());
+  constexpr int kDtxPeriod = 400;
+  nack.UpdateLastReceivedPacket(seq_num + 2,
+                                timestamp + kDtxPeriod * kSampleRateHz / 1000);
+  EXPECT_TRUE(nack.GetNackList(0).empty());
+}
+
+TEST(NackTrackerTest, DoNotNackIfLossRateIsTooHigh) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-NetEqNackTrackerConfig/max_loss_rate:0.4/");
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  auto add_packet = [&nack, &seq_num, &timestamp](bool received) {
+    if (received) {
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    }
+    seq_num++;
+    timestamp += kTimestampIncrement;
+  };
+  for (int i = 0; i < 500; i++) {
+    add_packet(true);
+    add_packet(false);
+  }
+  // Expect 50% loss rate which is higher that the configured maximum 40%.
+  EXPECT_NEAR(nack.GetPacketLossRateForTest(), 1 << 29, (1 << 30) / 100);
+  EXPECT_TRUE(nack.GetNackList(0).empty());
+}
+
+TEST(NackTrackerTest, OnlyNackIfRttIsValid) {
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-NetEqNackTrackerConfig/require_valid_rtt:true/");
+  const int kNackListSize = 200;
+  NackTracker nack;
+  nack.UpdateSampleRate(kSampleRateHz);
+  nack.SetMaxNackListSize(kNackListSize);
+  uint16_t seq_num = 0;
+  uint32_t timestamp = 0x87654321;
+  auto add_packet = [&nack, &seq_num, &timestamp](bool received) {
+    if (received) {
+      nack.UpdateLastReceivedPacket(seq_num, timestamp);
+    }
+    seq_num++;
+    timestamp += kTimestampIncrement;
+  };
+  add_packet(true);
+  add_packet(false);
+  add_packet(true);
+  EXPECT_TRUE(nack.GetNackList(0).empty());
+  EXPECT_FALSE(nack.GetNackList(10).empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
new file mode 100644
index 0000000000..cf310d1efb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_decoder_plc_unittest.cc
@@ -0,0 +1,313 @@
+/*
+ *  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.
+ */
+
+// Test to verify correct operation when using the decoder-internal PLC.
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/audio_checksum.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/encode_neteq_input.h"
+#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/audio_decoder_proxy_factory.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr int kSampleRateHz = 32000;
+constexpr int kRunTimeMs = 10000;
+
+// This class implements a fake decoder. The decoder will read audio from a file
+// and present as output, both for regular decoding and for PLC.
+class AudioDecoderPlc : public AudioDecoder {
+ public:
+  AudioDecoderPlc(std::unique_ptr<InputAudioFile> input, int sample_rate_hz)
+      : input_(std::move(input)), sample_rate_hz_(sample_rate_hz) {}
+
+  void Reset() override {}
+  int SampleRateHz() const override { return sample_rate_hz_; }
+  size_t Channels() const override { return 1; }
+  int DecodeInternal(const uint8_t* /*encoded*/,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override {
+    RTC_CHECK_GE(encoded_len / 2, 10 * sample_rate_hz_ / 1000);
+    RTC_CHECK_LE(encoded_len / 2, 2 * 10 * sample_rate_hz_ / 1000);
+    RTC_CHECK_EQ(sample_rate_hz, sample_rate_hz_);
+    RTC_CHECK(decoded);
+    RTC_CHECK(speech_type);
+    RTC_CHECK(input_->Read(encoded_len / 2, decoded));
+    *speech_type = kSpeech;
+    last_was_plc_ = false;
+    return encoded_len / 2;
+  }
+
+  void GeneratePlc(size_t requested_samples_per_channel,
+                   rtc::BufferT<int16_t>* concealment_audio) override {
+    // Instead of generating random data for GeneratePlc we use the same data as
+    // the input, so we can check that we produce the same result independently
+    // of the losses.
+    RTC_DCHECK_EQ(requested_samples_per_channel, 10 * sample_rate_hz_ / 1000);
+
+    // Must keep a local copy of this since DecodeInternal sets it to false.
+    const bool last_was_plc = last_was_plc_;
+
+    std::vector<int16_t> decoded(5760);
+    SpeechType speech_type;
+    int dec_len = DecodeInternal(nullptr, 2 * 10 * sample_rate_hz_ / 1000,
+                                 sample_rate_hz_, decoded.data(), &speech_type);
+    concealment_audio->AppendData(decoded.data(), dec_len);
+    concealed_samples_ += rtc::checked_cast<size_t>(dec_len);
+
+    if (!last_was_plc) {
+      ++concealment_events_;
+    }
+    last_was_plc_ = true;
+  }
+
+  size_t concealed_samples() { return concealed_samples_; }
+  size_t concealment_events() { return concealment_events_; }
+
+ private:
+  const std::unique_ptr<InputAudioFile> input_;
+  const int sample_rate_hz_;
+  size_t concealed_samples_ = 0;
+  size_t concealment_events_ = 0;
+  bool last_was_plc_ = false;
+};
+
+// An input sample generator which generates only zero-samples.
+class ZeroSampleGenerator : public EncodeNetEqInput::Generator {
+ public:
+  rtc::ArrayView<const int16_t> Generate(size_t num_samples) override {
+    vec.resize(num_samples, 0);
+    rtc::ArrayView<const int16_t> view(vec);
+    RTC_DCHECK_EQ(view.size(), num_samples);
+    return view;
+  }
+
+ private:
+  std::vector<int16_t> vec;
+};
+
+// A NetEqInput which connects to another NetEqInput, but drops a number of
+// consecutive packets on the way
+class LossyInput : public NetEqInput {
+ public:
+  LossyInput(int loss_cadence,
+             int burst_length,
+             std::unique_ptr<NetEqInput> input)
+      : loss_cadence_(loss_cadence),
+        burst_length_(burst_length),
+        input_(std::move(input)) {}
+
+  absl::optional<int64_t> NextPacketTime() const override {
+    return input_->NextPacketTime();
+  }
+
+  absl::optional<int64_t> NextOutputEventTime() const override {
+    return input_->NextOutputEventTime();
+  }
+
+  std::unique_ptr<PacketData> PopPacket() override {
+    if (loss_cadence_ != 0 && (++count_ % loss_cadence_) == 0) {
+      // Pop `burst_length_` packets to create the loss.
+      auto packet_to_return = input_->PopPacket();
+      for (int i = 0; i < burst_length_; i++) {
+        input_->PopPacket();
+      }
+      return packet_to_return;
+    }
+    return input_->PopPacket();
+  }
+
+  void AdvanceOutputEvent() override { return input_->AdvanceOutputEvent(); }
+
+  bool ended() const override { return input_->ended(); }
+
+  absl::optional<RTPHeader> NextHeader() const override {
+    return input_->NextHeader();
+  }
+
+ private:
+  const int loss_cadence_;
+  const int burst_length_;
+  int count_ = 0;
+  const std::unique_ptr<NetEqInput> input_;
+};
+
+class AudioChecksumWithOutput : public AudioChecksum {
+ public:
+  explicit AudioChecksumWithOutput(std::string* output_str)
+      : output_str_(*output_str) {}
+  ~AudioChecksumWithOutput() { output_str_ = Finish(); }
+
+ private:
+  std::string& output_str_;
+};
+
+struct TestStatistics {
+  NetEqNetworkStatistics network;
+  NetEqLifetimeStatistics lifetime;
+};
+
+TestStatistics RunTest(int loss_cadence,
+                       int burst_length,
+                       std::string* checksum) {
+  NetEq::Config config;
+  config.for_test_no_time_stretching = true;
+
+  // The input is mostly useless. It sends zero-samples to a PCM16b encoder,
+  // but the actual encoded samples will never be used by the decoder in the
+  // test. See below about the decoder.
+  auto generator = std::make_unique<ZeroSampleGenerator>();
+  constexpr int kPayloadType = 100;
+  AudioEncoderPcm16B::Config encoder_config;
+  encoder_config.sample_rate_hz = kSampleRateHz;
+  encoder_config.payload_type = kPayloadType;
+  auto encoder = std::make_unique<AudioEncoderPcm16B>(encoder_config);
+  auto input = std::make_unique<EncodeNetEqInput>(
+      std::move(generator), std::move(encoder), kRunTimeMs);
+  // Wrap the input in a loss function.
+  auto lossy_input = std::make_unique<LossyInput>(loss_cadence, burst_length,
+                                                  std::move(input));
+
+  // Setting up decoders.
+  NetEqTest::DecoderMap decoders;
+  // Using a fake decoder which simply reads the output audio from a file.
+  auto input_file = std::make_unique<InputAudioFile>(
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"));
+  AudioDecoderPlc dec(std::move(input_file), kSampleRateHz);
+  // Masquerading as a PCM16b decoder.
+  decoders.emplace(kPayloadType, SdpAudioFormat("l16", 32000, 1));
+
+  // Output is simply a checksum calculator.
+  auto output = std::make_unique<AudioChecksumWithOutput>(checksum);
+
+  // No callback objects.
+  NetEqTest::Callbacks callbacks;
+
+  NetEqTest neteq_test(
+      config, /*decoder_factory=*/
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&dec),
+      /*codecs=*/decoders, /*text_log=*/nullptr, /*neteq_factory=*/nullptr,
+      /*input=*/std::move(lossy_input), std::move(output), callbacks);
+  EXPECT_LE(kRunTimeMs, neteq_test.Run());
+
+  auto lifetime_stats = neteq_test.LifetimeStats();
+  EXPECT_EQ(dec.concealed_samples(), lifetime_stats.concealed_samples);
+  EXPECT_EQ(dec.concealment_events(), lifetime_stats.concealment_events);
+  return {neteq_test.SimulationStats(), neteq_test.LifetimeStats()};
+}
+}  // namespace
+
+// Check that some basic metrics are produced in the right direction. In
+// particular, expand_rate should only increase if there are losses present. Our
+// dummy decoder is designed such as the checksum should always be the same
+// regardless of the losses given that calls are executed in the right order.
+TEST(NetEqDecoderPlc, BasicMetrics) {
+  std::string checksum;
+
+  // Drop 1 packet every 10 packets.
+  auto stats = RunTest(10, 1, &checksum);
+
+  std::string checksum_no_loss;
+  auto stats_no_loss = RunTest(0, 0, &checksum_no_loss);
+
+  EXPECT_EQ(checksum, checksum_no_loss);
+
+  EXPECT_EQ(stats.network.preemptive_rate,
+            stats_no_loss.network.preemptive_rate);
+  EXPECT_EQ(stats.network.accelerate_rate,
+            stats_no_loss.network.accelerate_rate);
+  EXPECT_EQ(0, stats_no_loss.network.expand_rate);
+  EXPECT_GT(stats.network.expand_rate, 0);
+}
+
+// Checks that interruptions are not counted in small losses but they are
+// correctly counted in long interruptions.
+TEST(NetEqDecoderPlc, CountInterruptions) {
+  std::string checksum;
+  std::string checksum_2;
+  std::string checksum_3;
+
+  // Half of the packets lost but in short interruptions.
+  auto stats_no_interruptions = RunTest(1, 1, &checksum);
+  // One lost of 500 ms (250 packets).
+  auto stats_one_interruption = RunTest(200, 250, &checksum_2);
+  // Two losses of 250ms each (125 packets).
+  auto stats_two_interruptions = RunTest(125, 125, &checksum_3);
+
+  EXPECT_EQ(checksum, checksum_2);
+  EXPECT_EQ(checksum, checksum_3);
+  EXPECT_GT(stats_no_interruptions.network.expand_rate, 0);
+  EXPECT_EQ(stats_no_interruptions.lifetime.total_interruption_duration_ms, 0);
+  EXPECT_EQ(stats_no_interruptions.lifetime.interruption_count, 0);
+
+  EXPECT_GT(stats_one_interruption.network.expand_rate, 0);
+  EXPECT_EQ(stats_one_interruption.lifetime.total_interruption_duration_ms,
+            5000);
+  EXPECT_EQ(stats_one_interruption.lifetime.interruption_count, 1);
+
+  EXPECT_GT(stats_two_interruptions.network.expand_rate, 0);
+  EXPECT_EQ(stats_two_interruptions.lifetime.total_interruption_duration_ms,
+            5000);
+  EXPECT_EQ(stats_two_interruptions.lifetime.interruption_count, 2);
+}
+
+// Checks that small losses do not produce interruptions.
+TEST(NetEqDecoderPlc, NoInterruptionsInSmallLosses) {
+  std::string checksum_1;
+  std::string checksum_4;
+
+  auto stats_1 = RunTest(300, 1, &checksum_1);
+  auto stats_4 = RunTest(300, 4, &checksum_4);
+
+  EXPECT_EQ(checksum_1, checksum_4);
+
+  EXPECT_EQ(stats_1.lifetime.interruption_count, 0);
+  EXPECT_EQ(stats_1.lifetime.total_interruption_duration_ms, 0);
+  EXPECT_EQ(stats_1.lifetime.concealed_samples, 640u);  // 20ms of concealment.
+  EXPECT_EQ(stats_1.lifetime.concealment_events, 1u);   // in just one event.
+
+  EXPECT_EQ(stats_4.lifetime.interruption_count, 0);
+  EXPECT_EQ(stats_4.lifetime.total_interruption_duration_ms, 0);
+  EXPECT_EQ(stats_4.lifetime.concealed_samples, 2560u);  // 80ms of concealment.
+  EXPECT_EQ(stats_4.lifetime.concealment_events, 1u);    // in just one event.
+}
+
+// Checks that interruptions of different sizes report correct duration.
+TEST(NetEqDecoderPlc, InterruptionsReportCorrectSize) {
+  std::string checksum;
+
+  for (int burst_length = 5; burst_length < 10; burst_length++) {
+    auto stats = RunTest(300, burst_length, &checksum);
+    auto duration = stats.lifetime.total_interruption_duration_ms;
+    if (burst_length < 8) {
+      EXPECT_EQ(duration, 0);
+    } else {
+      EXPECT_EQ(duration, burst_length * 20);
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc
new file mode 100644
index 0000000000..6a6367d045
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -0,0 +1,2141 @@
+/*
+ *  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/neteq/neteq_impl.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <cstring>
+#include <list>
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/tick_timer.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "modules/audio_coding/neteq/accelerate.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/comfort_noise.h"
+#include "modules/audio_coding/neteq/decision_logic.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/dtmf_buffer.h"
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/merge.h"
+#include "modules/audio_coding/neteq/nack_tracker.h"
+#include "modules/audio_coding/neteq/normal.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "modules/audio_coding/neteq/post_decode_vad.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/time_stretch.h"
+#include "modules/audio_coding/neteq/timestamp_scaler.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/sanitizer.h"
+#include "rtc_base/strings/audio_format_to_string.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+
+std::unique_ptr<NetEqController> CreateNetEqController(
+    const NetEqControllerFactory& controller_factory,
+    int base_min_delay,
+    int max_packets_in_buffer,
+    bool allow_time_stretching,
+    TickTimer* tick_timer,
+    webrtc::Clock* clock) {
+  NetEqController::Config config;
+  config.base_min_delay_ms = base_min_delay;
+  config.max_packets_in_buffer = max_packets_in_buffer;
+  config.allow_time_stretching = allow_time_stretching;
+  config.tick_timer = tick_timer;
+  config.clock = clock;
+  return controller_factory.CreateNetEqController(config);
+}
+
+}  // namespace
+
+NetEqImpl::Dependencies::Dependencies(
+    const NetEq::Config& config,
+    Clock* clock,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+    const NetEqControllerFactory& controller_factory)
+    : clock(clock),
+      tick_timer(new TickTimer),
+      stats(new StatisticsCalculator),
+      decoder_database(
+          new DecoderDatabase(decoder_factory, config.codec_pair_id)),
+      dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)),
+      dtmf_tone_generator(new DtmfToneGenerator),
+      packet_buffer(
+          new PacketBuffer(config.max_packets_in_buffer, tick_timer.get())),
+      neteq_controller(
+          CreateNetEqController(controller_factory,
+                                config.min_delay_ms,
+                                config.max_packets_in_buffer,
+                                !config.for_test_no_time_stretching,
+                                tick_timer.get(),
+                                clock)),
+      red_payload_splitter(new RedPayloadSplitter),
+      timestamp_scaler(new TimestampScaler(*decoder_database)),
+      accelerate_factory(new AccelerateFactory),
+      expand_factory(new ExpandFactory),
+      preemptive_expand_factory(new PreemptiveExpandFactory) {}
+
+NetEqImpl::Dependencies::~Dependencies() = default;
+
+NetEqImpl::NetEqImpl(const NetEq::Config& config,
+                     Dependencies&& deps,
+                     bool create_components)
+    : clock_(deps.clock),
+      tick_timer_(std::move(deps.tick_timer)),
+      decoder_database_(std::move(deps.decoder_database)),
+      dtmf_buffer_(std::move(deps.dtmf_buffer)),
+      dtmf_tone_generator_(std::move(deps.dtmf_tone_generator)),
+      packet_buffer_(std::move(deps.packet_buffer)),
+      red_payload_splitter_(std::move(deps.red_payload_splitter)),
+      timestamp_scaler_(std::move(deps.timestamp_scaler)),
+      vad_(new PostDecodeVad()),
+      expand_factory_(std::move(deps.expand_factory)),
+      accelerate_factory_(std::move(deps.accelerate_factory)),
+      preemptive_expand_factory_(std::move(deps.preemptive_expand_factory)),
+      stats_(std::move(deps.stats)),
+      controller_(std::move(deps.neteq_controller)),
+      last_mode_(Mode::kNormal),
+      decoded_buffer_length_(kMaxFrameSize),
+      decoded_buffer_(new int16_t[decoded_buffer_length_]),
+      playout_timestamp_(0),
+      new_codec_(false),
+      timestamp_(0),
+      reset_decoder_(false),
+      first_packet_(true),
+      enable_fast_accelerate_(config.enable_fast_accelerate),
+      nack_enabled_(false),
+      enable_muted_state_(config.enable_muted_state),
+      expand_uma_logger_("WebRTC.Audio.ExpandRatePercent",
+                         10,  // Report once every 10 s.
+                         tick_timer_.get()),
+      speech_expand_uma_logger_("WebRTC.Audio.SpeechExpandRatePercent",
+                                10,  // Report once every 10 s.
+                                tick_timer_.get()),
+      no_time_stretching_(config.for_test_no_time_stretching) {
+  RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
+  int fs = config.sample_rate_hz;
+  if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
+    RTC_LOG(LS_ERROR) << "Sample rate " << fs
+                      << " Hz not supported. "
+                         "Changing to 8000 Hz.";
+    fs = 8000;
+  }
+  controller_->SetMaximumDelay(config.max_delay_ms);
+  fs_hz_ = fs;
+  fs_mult_ = fs / 8000;
+  last_output_sample_rate_hz_ = fs;
+  output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
+  controller_->SetSampleRate(fs_hz_, output_size_samples_);
+  decoder_frame_length_ = 2 * output_size_samples_;  // 20 ms.
+  if (create_components) {
+    SetSampleRateAndChannels(fs, 1);  // Default is 1 channel.
+  }
+  RTC_DCHECK(!vad_->enabled());
+  if (config.enable_post_decode_vad) {
+    vad_->Enable();
+  }
+}
+
+NetEqImpl::~NetEqImpl() = default;
+
+int NetEqImpl::InsertPacket(const RTPHeader& rtp_header,
+                            rtc::ArrayView<const uint8_t> payload) {
+  rtc::MsanCheckInitialized(payload);
+  TRACE_EVENT0("webrtc", "NetEqImpl::InsertPacket");
+  MutexLock lock(&mutex_);
+  if (InsertPacketInternal(rtp_header, payload) != 0) {
+    return kFail;
+  }
+  return kOK;
+}
+
+void NetEqImpl::InsertEmptyPacket(const RTPHeader& rtp_header) {
+  MutexLock lock(&mutex_);
+  if (nack_enabled_) {
+    nack_->UpdateLastReceivedPacket(rtp_header.sequenceNumber,
+                                    rtp_header.timestamp);
+  }
+  controller_->RegisterEmptyPacket();
+}
+
+namespace {
+void SetAudioFrameActivityAndType(bool vad_enabled,
+                                  NetEqImpl::OutputType type,
+                                  AudioFrame::VADActivity last_vad_activity,
+                                  AudioFrame* audio_frame) {
+  switch (type) {
+    case NetEqImpl::OutputType::kNormalSpeech: {
+      audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+      audio_frame->vad_activity_ = AudioFrame::kVadActive;
+      break;
+    }
+    case NetEqImpl::OutputType::kVadPassive: {
+      // This should only be reached if the VAD is enabled.
+      RTC_DCHECK(vad_enabled);
+      audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
+      audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+      break;
+    }
+    case NetEqImpl::OutputType::kCNG: {
+      audio_frame->speech_type_ = AudioFrame::kCNG;
+      audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+      break;
+    }
+    case NetEqImpl::OutputType::kPLC: {
+      audio_frame->speech_type_ = AudioFrame::kPLC;
+      audio_frame->vad_activity_ = last_vad_activity;
+      break;
+    }
+    case NetEqImpl::OutputType::kPLCCNG: {
+      audio_frame->speech_type_ = AudioFrame::kPLCCNG;
+      audio_frame->vad_activity_ = AudioFrame::kVadPassive;
+      break;
+    }
+    case NetEqImpl::OutputType::kCodecPLC: {
+      audio_frame->speech_type_ = AudioFrame::kCodecPLC;
+      audio_frame->vad_activity_ = last_vad_activity;
+      break;
+    }
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  if (!vad_enabled) {
+    // Always set kVadUnknown when receive VAD is inactive.
+    audio_frame->vad_activity_ = AudioFrame::kVadUnknown;
+  }
+}
+}  // namespace
+
+int NetEqImpl::GetAudio(AudioFrame* audio_frame,
+                        bool* muted,
+                        int* current_sample_rate_hz,
+                        absl::optional<Operation> action_override) {
+  TRACE_EVENT0("webrtc", "NetEqImpl::GetAudio");
+  MutexLock lock(&mutex_);
+  if (GetAudioInternal(audio_frame, muted, action_override) != 0) {
+    return kFail;
+  }
+  RTC_DCHECK_EQ(
+      audio_frame->sample_rate_hz_,
+      rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
+  RTC_DCHECK_EQ(*muted, audio_frame->muted());
+  SetAudioFrameActivityAndType(vad_->enabled(), LastOutputType(),
+                               last_vad_activity_, audio_frame);
+  last_vad_activity_ = audio_frame->vad_activity_;
+  last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_;
+  RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
+             last_output_sample_rate_hz_ == 16000 ||
+             last_output_sample_rate_hz_ == 32000 ||
+             last_output_sample_rate_hz_ == 48000)
+      << "Unexpected sample rate " << last_output_sample_rate_hz_;
+
+  if (current_sample_rate_hz) {
+    *current_sample_rate_hz = last_output_sample_rate_hz_;
+  }
+
+  return kOK;
+}
+
+void NetEqImpl::SetCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+  MutexLock lock(&mutex_);
+  const std::vector<int> changed_payload_types =
+      decoder_database_->SetCodecs(codecs);
+  for (const int pt : changed_payload_types) {
+    packet_buffer_->DiscardPacketsWithPayloadType(pt, stats_.get());
+  }
+}
+
+bool NetEqImpl::RegisterPayloadType(int rtp_payload_type,
+                                    const SdpAudioFormat& audio_format) {
+  RTC_LOG(LS_VERBOSE) << "NetEqImpl::RegisterPayloadType: payload type "
+                      << rtp_payload_type << ", codec "
+                      << rtc::ToString(audio_format);
+  MutexLock lock(&mutex_);
+  return decoder_database_->RegisterPayload(rtp_payload_type, audio_format) ==
+         DecoderDatabase::kOK;
+}
+
+int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) {
+  MutexLock lock(&mutex_);
+  int ret = decoder_database_->Remove(rtp_payload_type);
+  if (ret == DecoderDatabase::kOK || ret == DecoderDatabase::kDecoderNotFound) {
+    packet_buffer_->DiscardPacketsWithPayloadType(rtp_payload_type,
+                                                  stats_.get());
+    return kOK;
+  }
+  return kFail;
+}
+
+void NetEqImpl::RemoveAllPayloadTypes() {
+  MutexLock lock(&mutex_);
+  decoder_database_->RemoveAll();
+}
+
+bool NetEqImpl::SetMinimumDelay(int delay_ms) {
+  MutexLock lock(&mutex_);
+  if (delay_ms >= 0 && delay_ms <= 10000) {
+    RTC_DCHECK(controller_.get());
+    return controller_->SetMinimumDelay(delay_ms);
+  }
+  return false;
+}
+
+bool NetEqImpl::SetMaximumDelay(int delay_ms) {
+  MutexLock lock(&mutex_);
+  if (delay_ms >= 0 && delay_ms <= 10000) {
+    RTC_DCHECK(controller_.get());
+    return controller_->SetMaximumDelay(delay_ms);
+  }
+  return false;
+}
+
+bool NetEqImpl::SetBaseMinimumDelayMs(int delay_ms) {
+  MutexLock lock(&mutex_);
+  if (delay_ms >= 0 && delay_ms <= 10000) {
+    return controller_->SetBaseMinimumDelay(delay_ms);
+  }
+  return false;
+}
+
+int NetEqImpl::GetBaseMinimumDelayMs() const {
+  MutexLock lock(&mutex_);
+  return controller_->GetBaseMinimumDelay();
+}
+
+int NetEqImpl::TargetDelayMs() const {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK(controller_.get());
+  return controller_->TargetLevelMs();
+}
+
+int NetEqImpl::FilteredCurrentDelayMs() const {
+  MutexLock lock(&mutex_);
+  // Sum up the filtered packet buffer level with the future length of the sync
+  // buffer.
+  const int delay_samples =
+      controller_->GetFilteredBufferLevel() + sync_buffer_->FutureLength();
+  // The division below will truncate. The return value is in ms.
+  return delay_samples / rtc::CheckedDivExact(fs_hz_, 1000);
+}
+
+int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK(decoder_database_.get());
+  *stats = CurrentNetworkStatisticsInternal();
+  stats_->GetNetworkStatistics(decoder_frame_length_, stats);
+  return 0;
+}
+
+NetEqNetworkStatistics NetEqImpl::CurrentNetworkStatistics() const {
+  MutexLock lock(&mutex_);
+  return CurrentNetworkStatisticsInternal();
+}
+
+NetEqNetworkStatistics NetEqImpl::CurrentNetworkStatisticsInternal() const {
+  RTC_DCHECK(decoder_database_.get());
+  NetEqNetworkStatistics stats;
+  const size_t total_samples_in_buffers =
+      packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) +
+      sync_buffer_->FutureLength();
+
+  RTC_DCHECK(controller_.get());
+  stats.preferred_buffer_size_ms = controller_->TargetLevelMs();
+  stats.jitter_peaks_found = controller_->PeakFound();
+  RTC_DCHECK_GT(fs_hz_, 0);
+  stats.current_buffer_size_ms =
+      static_cast<uint16_t>(total_samples_in_buffers * 1000 / fs_hz_);
+  return stats;
+}
+
+NetEqLifetimeStatistics NetEqImpl::GetLifetimeStatistics() const {
+  MutexLock lock(&mutex_);
+  return stats_->GetLifetimeStatistics();
+}
+
+NetEqOperationsAndState NetEqImpl::GetOperationsAndState() const {
+  MutexLock lock(&mutex_);
+  auto result = stats_->GetOperationsAndState();
+  result.current_buffer_size_ms =
+      (packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) +
+       sync_buffer_->FutureLength()) *
+      1000 / fs_hz_;
+  result.current_frame_size_ms = decoder_frame_length_ * 1000 / fs_hz_;
+  result.next_packet_available = packet_buffer_->PeekNextPacket() &&
+                                 packet_buffer_->PeekNextPacket()->timestamp ==
+                                     sync_buffer_->end_timestamp();
+  return result;
+}
+
+void NetEqImpl::EnableVad() {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK(vad_.get());
+  vad_->Enable();
+}
+
+void NetEqImpl::DisableVad() {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK(vad_.get());
+  vad_->Disable();
+}
+
+absl::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
+  MutexLock lock(&mutex_);
+  if (first_packet_ || last_mode_ == Mode::kRfc3389Cng ||
+      last_mode_ == Mode::kCodecInternalCng) {
+    // We don't have a valid RTP timestamp until we have decoded our first
+    // RTP packet. Also, the RTP timestamp is not accurate while playing CNG,
+    // which is indicated by returning an empty value.
+    return absl::nullopt;
+  }
+  return timestamp_scaler_->ToExternal(playout_timestamp_);
+}
+
+int NetEqImpl::last_output_sample_rate_hz() const {
+  MutexLock lock(&mutex_);
+  return last_output_sample_rate_hz_;
+}
+
+absl::optional<NetEq::DecoderFormat> NetEqImpl::GetDecoderFormat(
+    int payload_type) const {
+  MutexLock lock(&mutex_);
+  const DecoderDatabase::DecoderInfo* const di =
+      decoder_database_->GetDecoderInfo(payload_type);
+  if (di) {
+    const AudioDecoder* const decoder = di->GetDecoder();
+    // TODO(kwiberg): Why the special case for RED?
+    return DecoderFormat{
+        /*sample_rate_hz=*/di->IsRed() ? 8000 : di->SampleRateHz(),
+        /*num_channels=*/
+        decoder ? rtc::dchecked_cast<int>(decoder->Channels()) : 1,
+        /*sdp_format=*/di->GetFormat()};
+  } else {
+    // Payload type not registered.
+    return absl::nullopt;
+  }
+}
+
+void NetEqImpl::FlushBuffers() {
+  MutexLock lock(&mutex_);
+  RTC_LOG(LS_VERBOSE) << "FlushBuffers";
+  packet_buffer_->Flush(stats_.get());
+  RTC_DCHECK(sync_buffer_.get());
+  RTC_DCHECK(expand_.get());
+  sync_buffer_->Flush();
+  sync_buffer_->set_next_index(sync_buffer_->next_index() -
+                               expand_->overlap_length());
+  // Set to wait for new codec.
+  first_packet_ = true;
+}
+
+void NetEqImpl::EnableNack(size_t max_nack_list_size) {
+  MutexLock lock(&mutex_);
+  if (!nack_enabled_) {
+    nack_ = std::make_unique<NackTracker>();
+    nack_enabled_ = true;
+    nack_->UpdateSampleRate(fs_hz_);
+  }
+  nack_->SetMaxNackListSize(max_nack_list_size);
+}
+
+void NetEqImpl::DisableNack() {
+  MutexLock lock(&mutex_);
+  nack_.reset();
+  nack_enabled_ = false;
+}
+
+std::vector<uint16_t> NetEqImpl::GetNackList(int64_t round_trip_time_ms) const {
+  MutexLock lock(&mutex_);
+  if (!nack_enabled_) {
+    return std::vector<uint16_t>();
+  }
+  RTC_DCHECK(nack_.get());
+  return nack_->GetNackList(round_trip_time_ms);
+}
+
+int NetEqImpl::SyncBufferSizeMs() const {
+  MutexLock lock(&mutex_);
+  return rtc::dchecked_cast<int>(sync_buffer_->FutureLength() /
+                                 rtc::CheckedDivExact(fs_hz_, 1000));
+}
+
+const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
+  MutexLock lock(&mutex_);
+  return sync_buffer_.get();
+}
+
+NetEq::Operation NetEqImpl::last_operation_for_test() const {
+  MutexLock lock(&mutex_);
+  return last_operation_;
+}
+
+// Methods below this line are private.
+
+int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
+                                    rtc::ArrayView<const uint8_t> payload) {
+  if (payload.empty()) {
+    RTC_LOG_F(LS_ERROR) << "payload is empty";
+    return kInvalidPointer;
+  }
+
+  Timestamp receive_time = clock_->CurrentTime();
+  stats_->ReceivedPacket();
+
+  PacketList packet_list;
+  // Insert packet in a packet list.
+  packet_list.push_back([&rtp_header, &payload, &receive_time] {
+    // Convert to Packet.
+    Packet packet;
+    packet.payload_type = rtp_header.payloadType;
+    packet.sequence_number = rtp_header.sequenceNumber;
+    packet.timestamp = rtp_header.timestamp;
+    packet.payload.SetData(payload.data(), payload.size());
+    packet.packet_info = RtpPacketInfo(rtp_header, receive_time);
+    // Waiting time will be set upon inserting the packet in the buffer.
+    RTC_DCHECK(!packet.waiting_time);
+    return packet;
+  }());
+
+  bool update_sample_rate_and_channels = first_packet_;
+
+  if (update_sample_rate_and_channels) {
+    // Reset timestamp scaling.
+    timestamp_scaler_->Reset();
+  }
+
+  if (!decoder_database_->IsRed(rtp_header.payloadType)) {
+    // Scale timestamp to internal domain (only for some codecs).
+    timestamp_scaler_->ToInternal(&packet_list);
+  }
+
+  // Store these for later use, since the first packet may very well disappear
+  // before we need these values.
+  uint32_t main_timestamp = packet_list.front().timestamp;
+  uint8_t main_payload_type = packet_list.front().payload_type;
+  uint16_t main_sequence_number = packet_list.front().sequence_number;
+
+  // Reinitialize NetEq if it's needed (changed SSRC or first call).
+  if (update_sample_rate_and_channels) {
+    // Note: `first_packet_` will be cleared further down in this method, once
+    // the packet has been successfully inserted into the packet buffer.
+
+    // Flush the packet buffer and DTMF buffer.
+    packet_buffer_->Flush(stats_.get());
+    dtmf_buffer_->Flush();
+
+    // Update audio buffer timestamp.
+    sync_buffer_->IncreaseEndTimestamp(main_timestamp - timestamp_);
+
+    // Update codecs.
+    timestamp_ = main_timestamp;
+  }
+
+  if (nack_enabled_) {
+    RTC_DCHECK(nack_);
+    if (update_sample_rate_and_channels) {
+      nack_->Reset();
+    }
+    nack_->UpdateLastReceivedPacket(main_sequence_number, main_timestamp);
+  }
+
+  // Check for RED payload type, and separate payloads into several packets.
+  if (decoder_database_->IsRed(rtp_header.payloadType)) {
+    if (!red_payload_splitter_->SplitRed(&packet_list)) {
+      return kRedundancySplitError;
+    }
+    // Only accept a few RED payloads of the same type as the main data,
+    // DTMF events and CNG.
+    red_payload_splitter_->CheckRedPayloads(&packet_list, *decoder_database_);
+    if (packet_list.empty()) {
+      return kRedundancySplitError;
+    }
+  }
+
+  // Check payload types.
+  if (decoder_database_->CheckPayloadTypes(packet_list) ==
+      DecoderDatabase::kDecoderNotFound) {
+    return kUnknownRtpPayloadType;
+  }
+
+  RTC_DCHECK(!packet_list.empty());
+
+  // Update main_timestamp, if new packets appear in the list
+  // after RED splitting.
+  if (decoder_database_->IsRed(rtp_header.payloadType)) {
+    timestamp_scaler_->ToInternal(&packet_list);
+    main_timestamp = packet_list.front().timestamp;
+    main_payload_type = packet_list.front().payload_type;
+    main_sequence_number = packet_list.front().sequence_number;
+  }
+
+  // Process DTMF payloads. Cycle through the list of packets, and pick out any
+  // DTMF payloads found.
+  PacketList::iterator it = packet_list.begin();
+  while (it != packet_list.end()) {
+    const Packet& current_packet = (*it);
+    RTC_DCHECK(!current_packet.payload.empty());
+    if (decoder_database_->IsDtmf(current_packet.payload_type)) {
+      DtmfEvent event;
+      int ret = DtmfBuffer::ParseEvent(current_packet.timestamp,
+                                       current_packet.payload.data(),
+                                       current_packet.payload.size(), &event);
+      if (ret != DtmfBuffer::kOK) {
+        return kDtmfParsingError;
+      }
+      if (dtmf_buffer_->InsertEvent(event) != DtmfBuffer::kOK) {
+        return kDtmfInsertError;
+      }
+      it = packet_list.erase(it);
+    } else {
+      ++it;
+    }
+  }
+
+  PacketList parsed_packet_list;
+  bool is_dtx = false;
+  while (!packet_list.empty()) {
+    Packet& packet = packet_list.front();
+    const DecoderDatabase::DecoderInfo* info =
+        decoder_database_->GetDecoderInfo(packet.payload_type);
+    if (!info) {
+      RTC_LOG(LS_WARNING) << "SplitAudio unknown payload type";
+      return kUnknownRtpPayloadType;
+    }
+
+    if (info->IsComfortNoise()) {
+      // Carry comfort noise packets along.
+      parsed_packet_list.splice(parsed_packet_list.end(), packet_list,
+                                packet_list.begin());
+    } else {
+      const auto sequence_number = packet.sequence_number;
+      const auto payload_type = packet.payload_type;
+      const Packet::Priority original_priority = packet.priority;
+      const auto& packet_info = packet.packet_info;
+      auto packet_from_result = [&](AudioDecoder::ParseResult& result) {
+        Packet new_packet;
+        new_packet.sequence_number = sequence_number;
+        new_packet.payload_type = payload_type;
+        new_packet.timestamp = result.timestamp;
+        new_packet.priority.codec_level = result.priority;
+        new_packet.priority.red_level = original_priority.red_level;
+        new_packet.packet_info = packet_info;
+        new_packet.frame = std::move(result.frame);
+        return new_packet;
+      };
+
+      std::vector<AudioDecoder::ParseResult> results =
+          info->GetDecoder()->ParsePayload(std::move(packet.payload),
+                                           packet.timestamp);
+      if (results.empty()) {
+        packet_list.pop_front();
+      } else {
+        bool first = true;
+        for (auto& result : results) {
+          RTC_DCHECK(result.frame);
+          RTC_DCHECK_GE(result.priority, 0);
+          is_dtx = is_dtx || result.frame->IsDtxPacket();
+          if (first) {
+            // Re-use the node and move it to parsed_packet_list.
+            packet_list.front() = packet_from_result(result);
+            parsed_packet_list.splice(parsed_packet_list.end(), packet_list,
+                                      packet_list.begin());
+            first = false;
+          } else {
+            parsed_packet_list.push_back(packet_from_result(result));
+          }
+        }
+      }
+    }
+  }
+
+  // Calculate the number of primary (non-FEC/RED) packets.
+  const size_t number_of_primary_packets = std::count_if(
+      parsed_packet_list.begin(), parsed_packet_list.end(),
+      [](const Packet& in) { return in.priority.codec_level == 0; });
+  if (number_of_primary_packets < parsed_packet_list.size()) {
+    stats_->SecondaryPacketsReceived(parsed_packet_list.size() -
+                                     number_of_primary_packets);
+  }
+
+  // Insert packets in buffer.
+  const int target_level_ms = controller_->TargetLevelMs();
+  const int ret = packet_buffer_->InsertPacketList(
+      &parsed_packet_list, *decoder_database_, &current_rtp_payload_type_,
+      &current_cng_rtp_payload_type_, stats_.get(), decoder_frame_length_,
+      last_output_sample_rate_hz_, target_level_ms);
+  bool buffer_flush_occured = false;
+  if (ret == PacketBuffer::kFlushed) {
+    // Reset DSP timestamp etc. if packet buffer flushed.
+    new_codec_ = true;
+    update_sample_rate_and_channels = true;
+    buffer_flush_occured = true;
+  } else if (ret == PacketBuffer::kPartialFlush) {
+    // Forward sync buffer timestamp
+    timestamp_ = packet_buffer_->PeekNextPacket()->timestamp;
+    sync_buffer_->IncreaseEndTimestamp(timestamp_ -
+                                       sync_buffer_->end_timestamp());
+    buffer_flush_occured = true;
+  } else if (ret != PacketBuffer::kOK) {
+    return kOtherError;
+  }
+
+  if (first_packet_) {
+    first_packet_ = false;
+    // Update the codec on the next GetAudio call.
+    new_codec_ = true;
+  }
+
+  if (current_rtp_payload_type_) {
+    RTC_DCHECK(decoder_database_->GetDecoderInfo(*current_rtp_payload_type_))
+        << "Payload type " << static_cast<int>(*current_rtp_payload_type_)
+        << " is unknown where it shouldn't be";
+  }
+
+  if (update_sample_rate_and_channels && !packet_buffer_->Empty()) {
+    // We do not use `current_rtp_payload_type_` to |set payload_type|, but
+    // get the next RTP header from `packet_buffer_` to obtain the payload type.
+    // The reason for it is the following corner case. If NetEq receives a
+    // CNG packet with a sample rate different than the current CNG then it
+    // flushes its buffer, assuming send codec must have been changed. However,
+    // payload type of the hypothetically new send codec is not known.
+    const Packet* next_packet = packet_buffer_->PeekNextPacket();
+    RTC_DCHECK(next_packet);
+    const int payload_type = next_packet->payload_type;
+    size_t channels = 1;
+    if (!decoder_database_->IsComfortNoise(payload_type)) {
+      AudioDecoder* decoder = decoder_database_->GetDecoder(payload_type);
+      RTC_DCHECK(decoder);  // Payloads are already checked to be valid.
+      channels = decoder->Channels();
+    }
+    const DecoderDatabase::DecoderInfo* decoder_info =
+        decoder_database_->GetDecoderInfo(payload_type);
+    RTC_DCHECK(decoder_info);
+    if (decoder_info->SampleRateHz() != fs_hz_ ||
+        channels != algorithm_buffer_->Channels()) {
+      SetSampleRateAndChannels(decoder_info->SampleRateHz(), channels);
+    }
+    if (nack_enabled_) {
+      RTC_DCHECK(nack_);
+      // Update the sample rate even if the rate is not new, because of Reset().
+      nack_->UpdateSampleRate(fs_hz_);
+    }
+  }
+
+  const DecoderDatabase::DecoderInfo* dec_info =
+      decoder_database_->GetDecoderInfo(main_payload_type);
+  RTC_DCHECK(dec_info);  // Already checked that the payload type is known.
+
+  NetEqController::PacketArrivedInfo info;
+  info.is_cng_or_dtmf = dec_info->IsComfortNoise() || dec_info->IsDtmf();
+  info.packet_length_samples =
+      number_of_primary_packets * decoder_frame_length_;
+  info.main_timestamp = main_timestamp;
+  info.main_sequence_number = main_sequence_number;
+  info.is_dtx = is_dtx;
+  info.buffer_flush = buffer_flush_occured;
+
+  const bool should_update_stats = !new_codec_;
+  auto relative_delay =
+      controller_->PacketArrived(fs_hz_, should_update_stats, info);
+  if (relative_delay) {
+    stats_->RelativePacketArrivalDelay(relative_delay.value());
+  }
+  return 0;
+}
+
+int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
+                                bool* muted,
+                                absl::optional<Operation> action_override) {
+  PacketList packet_list;
+  DtmfEvent dtmf_event;
+  Operation operation;
+  bool play_dtmf;
+  *muted = false;
+  last_decoded_packet_infos_.clear();
+  tick_timer_->Increment();
+  stats_->IncreaseCounter(output_size_samples_, fs_hz_);
+  const auto lifetime_stats = stats_->GetLifetimeStatistics();
+  expand_uma_logger_.UpdateSampleCounter(lifetime_stats.concealed_samples,
+                                         fs_hz_);
+  speech_expand_uma_logger_.UpdateSampleCounter(
+      lifetime_stats.concealed_samples -
+          lifetime_stats.silent_concealed_samples,
+      fs_hz_);
+
+  // Check for muted state.
+  if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) {
+    RTC_DCHECK_EQ(last_mode_, Mode::kExpand);
+    audio_frame->Reset();
+    RTC_DCHECK(audio_frame->muted());  // Reset() should mute the frame.
+    playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
+    audio_frame->sample_rate_hz_ = fs_hz_;
+    // Make sure the total number of samples fits in the AudioFrame.
+    if (output_size_samples_ * sync_buffer_->Channels() >
+        AudioFrame::kMaxDataSizeSamples) {
+      return kSampleUnderrun;
+    }
+    audio_frame->samples_per_channel_ = output_size_samples_;
+    audio_frame->timestamp_ =
+        first_packet_
+            ? 0
+            : timestamp_scaler_->ToExternal(playout_timestamp_) -
+                  static_cast<uint32_t>(audio_frame->samples_per_channel_);
+    audio_frame->num_channels_ = sync_buffer_->Channels();
+    stats_->ExpandedNoiseSamples(output_size_samples_, false);
+    controller_->NotifyMutedState();
+    *muted = true;
+    return 0;
+  }
+  int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
+                                 &play_dtmf, action_override);
+  if (return_value != 0) {
+    last_mode_ = Mode::kError;
+    return return_value;
+  }
+
+  AudioDecoder::SpeechType speech_type;
+  int length = 0;
+  const size_t start_num_packets = packet_list.size();
+  int decode_return_value =
+      Decode(&packet_list, &operation, &length, &speech_type);
+
+  RTC_DCHECK(vad_.get());
+  bool sid_frame_available =
+      (operation == Operation::kRfc3389Cng && !packet_list.empty());
+  vad_->Update(decoded_buffer_.get(), static_cast<size_t>(length), speech_type,
+               sid_frame_available, fs_hz_);
+
+  // This is the criterion that we did decode some data through the speech
+  // decoder, and the operation resulted in comfort noise.
+  const bool codec_internal_sid_frame =
+      (speech_type == AudioDecoder::kComfortNoise &&
+       start_num_packets > packet_list.size());
+
+  if (sid_frame_available || codec_internal_sid_frame) {
+    // Start a new stopwatch since we are decoding a new CNG packet.
+    generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch();
+  }
+
+  algorithm_buffer_->Clear();
+  switch (operation) {
+    case Operation::kNormal: {
+      DoNormal(decoded_buffer_.get(), length, speech_type, play_dtmf);
+      if (length > 0) {
+        stats_->DecodedOutputPlayed();
+      }
+      break;
+    }
+    case Operation::kMerge: {
+      DoMerge(decoded_buffer_.get(), length, speech_type, play_dtmf);
+      break;
+    }
+    case Operation::kExpand: {
+      RTC_DCHECK_EQ(return_value, 0);
+      if (!current_rtp_payload_type_ || !DoCodecPlc()) {
+        return_value = DoExpand(play_dtmf);
+      }
+      RTC_DCHECK_GE(sync_buffer_->FutureLength() - expand_->overlap_length(),
+                    output_size_samples_);
+      break;
+    }
+    case Operation::kAccelerate:
+    case Operation::kFastAccelerate: {
+      const bool fast_accelerate =
+          enable_fast_accelerate_ && (operation == Operation::kFastAccelerate);
+      return_value = DoAccelerate(decoded_buffer_.get(), length, speech_type,
+                                  play_dtmf, fast_accelerate);
+      break;
+    }
+    case Operation::kPreemptiveExpand: {
+      return_value = DoPreemptiveExpand(decoded_buffer_.get(), length,
+                                        speech_type, play_dtmf);
+      break;
+    }
+    case Operation::kRfc3389Cng:
+    case Operation::kRfc3389CngNoPacket: {
+      return_value = DoRfc3389Cng(&packet_list, play_dtmf);
+      break;
+    }
+    case Operation::kCodecInternalCng: {
+      // This handles the case when there is no transmission and the decoder
+      // should produce internal comfort noise.
+      // TODO(hlundin): Write test for codec-internal CNG.
+      DoCodecInternalCng(decoded_buffer_.get(), length);
+      break;
+    }
+    case Operation::kDtmf: {
+      // TODO(hlundin): Write test for this.
+      return_value = DoDtmf(dtmf_event, &play_dtmf);
+      break;
+    }
+    case Operation::kUndefined: {
+      RTC_LOG(LS_ERROR) << "Invalid operation kUndefined.";
+      RTC_DCHECK_NOTREACHED();  // This should not happen.
+      last_mode_ = Mode::kError;
+      return kInvalidOperation;
+    }
+  }  // End of switch.
+  last_operation_ = operation;
+  if (return_value < 0) {
+    return return_value;
+  }
+
+  if (last_mode_ != Mode::kRfc3389Cng) {
+    comfort_noise_->Reset();
+  }
+
+  // We treat it as if all packets referenced to by `last_decoded_packet_infos_`
+  // were mashed together when creating the samples in `algorithm_buffer_`.
+  RtpPacketInfos packet_infos(last_decoded_packet_infos_);
+
+  // Copy samples from `algorithm_buffer_` to `sync_buffer_`.
+  //
+  // TODO(bugs.webrtc.org/10757):
+  //   We would in the future also like to pass `packet_infos` so that we can do
+  //   sample-perfect tracking of that information across `sync_buffer_`.
+  sync_buffer_->PushBack(*algorithm_buffer_);
+
+  // Extract data from `sync_buffer_` to `output`.
+  size_t num_output_samples_per_channel = output_size_samples_;
+  size_t num_output_samples = output_size_samples_ * sync_buffer_->Channels();
+  if (num_output_samples > AudioFrame::kMaxDataSizeSamples) {
+    RTC_LOG(LS_WARNING) << "Output array is too short. "
+                        << AudioFrame::kMaxDataSizeSamples << " < "
+                        << output_size_samples_ << " * "
+                        << sync_buffer_->Channels();
+    num_output_samples = AudioFrame::kMaxDataSizeSamples;
+    num_output_samples_per_channel =
+        AudioFrame::kMaxDataSizeSamples / sync_buffer_->Channels();
+  }
+  sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
+                                        audio_frame);
+  audio_frame->sample_rate_hz_ = fs_hz_;
+  // TODO(bugs.webrtc.org/10757):
+  //   We don't have the ability to properly track individual packets once their
+  //   audio samples have entered `sync_buffer_`. So for now, treat it as if
+  //   `packet_infos` from packets decoded by the current `GetAudioInternal()`
+  //   call were all consumed assembling the current audio frame and the current
+  //   audio frame only.
+  audio_frame->packet_infos_ = std::move(packet_infos);
+  if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
+    // The sync buffer should always contain `overlap_length` samples, but now
+    // too many samples have been extracted. Reinstall the `overlap_length`
+    // lookahead by moving the index.
+    const size_t missing_lookahead_samples =
+        expand_->overlap_length() - sync_buffer_->FutureLength();
+    RTC_DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples);
+    sync_buffer_->set_next_index(sync_buffer_->next_index() -
+                                 missing_lookahead_samples);
+  }
+  if (audio_frame->samples_per_channel_ != output_size_samples_) {
+    RTC_LOG(LS_ERROR) << "audio_frame->samples_per_channel_ ("
+                      << audio_frame->samples_per_channel_
+                      << ") != output_size_samples_ (" << output_size_samples_
+                      << ")";
+    // TODO(minyue): treatment of under-run, filling zeros
+    audio_frame->Mute();
+    return kSampleUnderrun;
+  }
+
+  // Should always have overlap samples left in the `sync_buffer_`.
+  RTC_DCHECK_GE(sync_buffer_->FutureLength(), expand_->overlap_length());
+
+  // TODO(yujo): For muted frames, this can be a copy rather than an addition.
+  if (play_dtmf) {
+    return_value = DtmfOverdub(dtmf_event, sync_buffer_->Channels(),
+                               audio_frame->mutable_data());
+  }
+
+  // Update the background noise parameters if last operation wrote data
+  // straight from the decoder to the `sync_buffer_`. That is, none of the
+  // operations that modify the signal can be followed by a parameter update.
+  if ((last_mode_ == Mode::kNormal) || (last_mode_ == Mode::kAccelerateFail) ||
+      (last_mode_ == Mode::kPreemptiveExpandFail) ||
+      (last_mode_ == Mode::kRfc3389Cng) ||
+      (last_mode_ == Mode::kCodecInternalCng)) {
+    background_noise_->Update(*sync_buffer_, *vad_.get());
+  }
+
+  if (operation == Operation::kDtmf) {
+    // DTMF data was written the end of `sync_buffer_`.
+    // Update index to end of DTMF data in `sync_buffer_`.
+    sync_buffer_->set_dtmf_index(sync_buffer_->Size());
+  }
+
+  if (last_mode_ != Mode::kExpand && last_mode_ != Mode::kCodecPlc) {
+    // If last operation was not expand, calculate the `playout_timestamp_` from
+    // the `sync_buffer_`. However, do not update the `playout_timestamp_` if it
+    // would be moved "backwards".
+    uint32_t temp_timestamp =
+        sync_buffer_->end_timestamp() -
+        static_cast<uint32_t>(sync_buffer_->FutureLength());
+    if (static_cast<int32_t>(temp_timestamp - playout_timestamp_) > 0) {
+      playout_timestamp_ = temp_timestamp;
+    }
+  } else {
+    // Use dead reckoning to estimate the `playout_timestamp_`.
+    playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
+  }
+  // Set the timestamp in the audio frame to zero before the first packet has
+  // been inserted. Otherwise, subtract the frame size in samples to get the
+  // timestamp of the first sample in the frame (playout_timestamp_ is the
+  // last + 1).
+  audio_frame->timestamp_ =
+      first_packet_
+          ? 0
+          : timestamp_scaler_->ToExternal(playout_timestamp_) -
+                static_cast<uint32_t>(audio_frame->samples_per_channel_);
+
+  if (!(last_mode_ == Mode::kRfc3389Cng ||
+        last_mode_ == Mode::kCodecInternalCng || last_mode_ == Mode::kExpand ||
+        last_mode_ == Mode::kCodecPlc)) {
+    generated_noise_stopwatch_.reset();
+  }
+
+  if (decode_return_value)
+    return decode_return_value;
+  return return_value;
+}
+
+int NetEqImpl::GetDecision(Operation* operation,
+                           PacketList* packet_list,
+                           DtmfEvent* dtmf_event,
+                           bool* play_dtmf,
+                           absl::optional<Operation> action_override) {
+  // Initialize output variables.
+  *play_dtmf = false;
+  *operation = Operation::kUndefined;
+
+  RTC_DCHECK(sync_buffer_.get());
+  uint32_t end_timestamp = sync_buffer_->end_timestamp();
+  if (!new_codec_) {
+    const uint32_t five_seconds_samples = 5 * fs_hz_;
+    packet_buffer_->DiscardOldPackets(end_timestamp, five_seconds_samples,
+                                      stats_.get());
+  }
+  const Packet* packet = packet_buffer_->PeekNextPacket();
+
+  RTC_DCHECK(!generated_noise_stopwatch_ ||
+             generated_noise_stopwatch_->ElapsedTicks() >= 1);
+  uint64_t generated_noise_samples =
+      generated_noise_stopwatch_ ? (generated_noise_stopwatch_->ElapsedTicks() -
+                                    1) * output_size_samples_ +
+                                       controller_->noise_fast_forward()
+                                 : 0;
+
+  if (controller_->CngRfc3389On() || last_mode_ == Mode::kRfc3389Cng) {
+    // Because of timestamp peculiarities, we have to "manually" disallow using
+    // a CNG packet with the same timestamp as the one that was last played.
+    // This can happen when using redundancy and will cause the timing to shift.
+    while (packet && decoder_database_->IsComfortNoise(packet->payload_type) &&
+           (end_timestamp >= packet->timestamp ||
+            end_timestamp + generated_noise_samples > packet->timestamp)) {
+      // Don't use this packet, discard it.
+      if (packet_buffer_->DiscardNextPacket(stats_.get()) !=
+          PacketBuffer::kOK) {
+        RTC_DCHECK_NOTREACHED();  // Must be ok by design.
+      }
+      // Check buffer again.
+      if (!new_codec_) {
+        packet_buffer_->DiscardOldPackets(end_timestamp, 5 * fs_hz_,
+                                          stats_.get());
+      }
+      packet = packet_buffer_->PeekNextPacket();
+    }
+  }
+
+  RTC_DCHECK(expand_.get());
+  const int samples_left = static_cast<int>(sync_buffer_->FutureLength() -
+                                            expand_->overlap_length());
+  if (last_mode_ == Mode::kAccelerateSuccess ||
+      last_mode_ == Mode::kAccelerateLowEnergy ||
+      last_mode_ == Mode::kPreemptiveExpandSuccess ||
+      last_mode_ == Mode::kPreemptiveExpandLowEnergy) {
+    // Subtract (samples_left + output_size_samples_) from sampleMemory.
+    controller_->AddSampleMemory(
+        -(samples_left + rtc::dchecked_cast<int>(output_size_samples_)));
+  }
+
+  // Check if it is time to play a DTMF event.
+  if (dtmf_buffer_->GetEvent(
+          static_cast<uint32_t>(end_timestamp + generated_noise_samples),
+          dtmf_event)) {
+    *play_dtmf = true;
+  }
+
+  // Get instruction.
+  RTC_DCHECK(sync_buffer_.get());
+  RTC_DCHECK(expand_.get());
+  generated_noise_samples =
+      generated_noise_stopwatch_
+          ? generated_noise_stopwatch_->ElapsedTicks() * output_size_samples_ +
+                controller_->noise_fast_forward()
+          : 0;
+  NetEqController::NetEqStatus status;
+  status.packet_buffer_info.dtx_or_cng =
+      packet_buffer_->ContainsDtxOrCngPacket(decoder_database_.get());
+  status.packet_buffer_info.num_samples =
+      packet_buffer_->NumSamplesInBuffer(decoder_frame_length_);
+  status.packet_buffer_info.span_samples = packet_buffer_->GetSpanSamples(
+      decoder_frame_length_, last_output_sample_rate_hz_, true);
+  status.packet_buffer_info.span_samples_no_dtx =
+      packet_buffer_->GetSpanSamples(decoder_frame_length_,
+                                     last_output_sample_rate_hz_, false);
+  status.packet_buffer_info.num_packets = packet_buffer_->NumPacketsInBuffer();
+  status.target_timestamp = sync_buffer_->end_timestamp();
+  status.expand_mutefactor = expand_->MuteFactor(0);
+  status.last_packet_samples = decoder_frame_length_;
+  status.last_mode = last_mode_;
+  status.play_dtmf = *play_dtmf;
+  status.generated_noise_samples = generated_noise_samples;
+  status.sync_buffer_samples = sync_buffer_->FutureLength();
+  if (packet) {
+    status.next_packet = {
+        packet->timestamp, packet->frame && packet->frame->IsDtxPacket(),
+        decoder_database_->IsComfortNoise(packet->payload_type)};
+  }
+  *operation = controller_->GetDecision(status, &reset_decoder_);
+
+  // Disallow time stretching if this packet is DTX, because such a decision may
+  // be based on earlier buffer level estimate, as we do not update buffer level
+  // during DTX. When we have a better way to update buffer level during DTX,
+  // this can be discarded.
+  if (packet && packet->frame && packet->frame->IsDtxPacket() &&
+      (*operation == Operation::kMerge ||
+       *operation == Operation::kAccelerate ||
+       *operation == Operation::kFastAccelerate ||
+       *operation == Operation::kPreemptiveExpand)) {
+    *operation = Operation::kNormal;
+  }
+
+  if (action_override) {
+    // Use the provided action instead of the decision NetEq decided on.
+    *operation = *action_override;
+  }
+  // Check if we already have enough samples in the `sync_buffer_`. If so,
+  // change decision to normal, unless the decision was merge, accelerate, or
+  // preemptive expand.
+  if (samples_left >= rtc::dchecked_cast<int>(output_size_samples_) &&
+      *operation != Operation::kMerge && *operation != Operation::kAccelerate &&
+      *operation != Operation::kFastAccelerate &&
+      *operation != Operation::kPreemptiveExpand) {
+    *operation = Operation::kNormal;
+    return 0;
+  }
+
+  controller_->ExpandDecision(*operation);
+  if ((last_mode_ == Mode::kCodecPlc) && (*operation != Operation::kExpand)) {
+    // Getting out of the PLC expand mode, reporting interruptions.
+    // NetEq PLC reports this metrics in expand.cc
+    stats_->EndExpandEvent(fs_hz_);
+  }
+
+  // Check conditions for reset.
+  if (new_codec_ || *operation == Operation::kUndefined) {
+    // The only valid reason to get kUndefined is that new_codec_ is set.
+    RTC_DCHECK(new_codec_);
+    if (*play_dtmf && !packet) {
+      timestamp_ = dtmf_event->timestamp;
+    } else {
+      if (!packet) {
+        RTC_LOG(LS_ERROR) << "Packet missing where it shouldn't.";
+        return -1;
+      }
+      timestamp_ = packet->timestamp;
+      if (*operation == Operation::kRfc3389CngNoPacket &&
+          decoder_database_->IsComfortNoise(packet->payload_type)) {
+        // Change decision to CNG packet, since we do have a CNG packet, but it
+        // was considered too early to use. Now, use it anyway.
+        *operation = Operation::kRfc3389Cng;
+      } else if (*operation != Operation::kRfc3389Cng) {
+        *operation = Operation::kNormal;
+      }
+    }
+    // Adjust `sync_buffer_` timestamp before setting `end_timestamp` to the
+    // new value.
+    sync_buffer_->IncreaseEndTimestamp(timestamp_ - end_timestamp);
+    end_timestamp = timestamp_;
+    new_codec_ = false;
+    controller_->SoftReset();
+    stats_->ResetMcu();
+  }
+
+  size_t required_samples = output_size_samples_;
+  const size_t samples_10_ms = static_cast<size_t>(80 * fs_mult_);
+  const size_t samples_20_ms = 2 * samples_10_ms;
+  const size_t samples_30_ms = 3 * samples_10_ms;
+
+  switch (*operation) {
+    case Operation::kExpand: {
+      timestamp_ = end_timestamp;
+      return 0;
+    }
+    case Operation::kRfc3389CngNoPacket:
+    case Operation::kCodecInternalCng: {
+      return 0;
+    }
+    case Operation::kDtmf: {
+      // TODO(hlundin): Write test for this.
+      // Update timestamp.
+      timestamp_ = end_timestamp;
+      const uint64_t generated_noise_samples =
+          generated_noise_stopwatch_
+              ? generated_noise_stopwatch_->ElapsedTicks() *
+                        output_size_samples_ +
+                    controller_->noise_fast_forward()
+              : 0;
+      if (generated_noise_samples > 0 && last_mode_ != Mode::kDtmf) {
+        // Make a jump in timestamp due to the recently played comfort noise.
+        uint32_t timestamp_jump =
+            static_cast<uint32_t>(generated_noise_samples);
+        sync_buffer_->IncreaseEndTimestamp(timestamp_jump);
+        timestamp_ += timestamp_jump;
+      }
+      return 0;
+    }
+    case Operation::kAccelerate:
+    case Operation::kFastAccelerate: {
+      // In order to do an accelerate we need at least 30 ms of audio data.
+      if (samples_left >= static_cast<int>(samples_30_ms)) {
+        // Already have enough data, so we do not need to extract any more.
+        controller_->set_sample_memory(samples_left);
+        controller_->set_prev_time_scale(true);
+        return 0;
+      } else if (samples_left >= static_cast<int>(samples_10_ms) &&
+                 decoder_frame_length_ >= samples_30_ms) {
+        // Avoid decoding more data as it might overflow the playout buffer.
+        *operation = Operation::kNormal;
+        return 0;
+      } else if (samples_left < static_cast<int>(samples_20_ms) &&
+                 decoder_frame_length_ < samples_30_ms) {
+        // Build up decoded data by decoding at least 20 ms of audio data. Do
+        // not perform accelerate yet, but wait until we only need to do one
+        // decoding.
+        required_samples = 2 * output_size_samples_;
+        *operation = Operation::kNormal;
+      }
+      // If none of the above is true, we have one of two possible situations:
+      // (1) 20 ms <= samples_left < 30 ms and decoder_frame_length_ < 30 ms; or
+      // (2) samples_left < 10 ms and decoder_frame_length_ >= 30 ms.
+      // In either case, we move on with the accelerate decision, and decode one
+      // frame now.
+      break;
+    }
+    case Operation::kPreemptiveExpand: {
+      // In order to do a preemptive expand we need at least 30 ms of decoded
+      // audio data.
+      if ((samples_left >= static_cast<int>(samples_30_ms)) ||
+          (samples_left >= static_cast<int>(samples_10_ms) &&
+           decoder_frame_length_ >= samples_30_ms)) {
+        // Already have enough data, so we do not need to extract any more.
+        // Or, avoid decoding more data as it might overflow the playout buffer.
+        // Still try preemptive expand, though.
+        controller_->set_sample_memory(samples_left);
+        controller_->set_prev_time_scale(true);
+        return 0;
+      }
+      if (samples_left < static_cast<int>(samples_20_ms) &&
+          decoder_frame_length_ < samples_30_ms) {
+        // Build up decoded data by decoding at least 20 ms of audio data.
+        // Still try to perform preemptive expand.
+        required_samples = 2 * output_size_samples_;
+      }
+      // Move on with the preemptive expand decision.
+      break;
+    }
+    case Operation::kMerge: {
+      required_samples =
+          std::max(merge_->RequiredFutureSamples(), required_samples);
+      break;
+    }
+    default: {
+      // Do nothing.
+    }
+  }
+
+  // Get packets from buffer.
+  int extracted_samples = 0;
+  if (packet) {
+    sync_buffer_->IncreaseEndTimestamp(packet->timestamp - end_timestamp);
+
+    if (*operation != Operation::kRfc3389Cng) {
+      // We are about to decode and use a non-CNG packet.
+      controller_->SetCngOff();
+    }
+
+    extracted_samples = ExtractPackets(required_samples, packet_list);
+    if (extracted_samples < 0) {
+      return kPacketBufferCorruption;
+    }
+  }
+
+  if (*operation == Operation::kAccelerate ||
+      *operation == Operation::kFastAccelerate ||
+      *operation == Operation::kPreemptiveExpand) {
+    controller_->set_sample_memory(samples_left + extracted_samples);
+    controller_->set_prev_time_scale(true);
+  }
+
+  if (*operation == Operation::kAccelerate ||
+      *operation == Operation::kFastAccelerate) {
+    // Check that we have enough data (30ms) to do accelerate.
+    if (extracted_samples + samples_left < static_cast<int>(samples_30_ms)) {
+      // TODO(hlundin): Write test for this.
+      // Not enough, do normal operation instead.
+      *operation = Operation::kNormal;
+    }
+  }
+
+  timestamp_ = sync_buffer_->end_timestamp();
+  return 0;
+}
+
+int NetEqImpl::Decode(PacketList* packet_list,
+                      Operation* operation,
+                      int* decoded_length,
+                      AudioDecoder::SpeechType* speech_type) {
+  *speech_type = AudioDecoder::kSpeech;
+
+  // When packet_list is empty, we may be in kCodecInternalCng mode, and for
+  // that we use current active decoder.
+  AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
+
+  if (!packet_list->empty()) {
+    const Packet& packet = packet_list->front();
+    uint8_t payload_type = packet.payload_type;
+    if (!decoder_database_->IsComfortNoise(payload_type)) {
+      decoder = decoder_database_->GetDecoder(payload_type);
+      RTC_DCHECK(decoder);
+      if (!decoder) {
+        RTC_LOG(LS_WARNING)
+            << "Unknown payload type " << static_cast<int>(payload_type);
+        packet_list->clear();
+        return kDecoderNotFound;
+      }
+      bool decoder_changed;
+      decoder_database_->SetActiveDecoder(payload_type, &decoder_changed);
+      if (decoder_changed) {
+        // We have a new decoder. Re-init some values.
+        const DecoderDatabase::DecoderInfo* decoder_info =
+            decoder_database_->GetDecoderInfo(payload_type);
+        RTC_DCHECK(decoder_info);
+        if (!decoder_info) {
+          RTC_LOG(LS_WARNING)
+              << "Unknown payload type " << static_cast<int>(payload_type);
+          packet_list->clear();
+          return kDecoderNotFound;
+        }
+        // If sampling rate or number of channels has changed, we need to make
+        // a reset.
+        if (decoder_info->SampleRateHz() != fs_hz_ ||
+            decoder->Channels() != algorithm_buffer_->Channels()) {
+          // TODO(tlegrand): Add unittest to cover this event.
+          SetSampleRateAndChannels(decoder_info->SampleRateHz(),
+                                   decoder->Channels());
+        }
+        sync_buffer_->set_end_timestamp(timestamp_);
+        playout_timestamp_ = timestamp_;
+      }
+    }
+  }
+
+  if (reset_decoder_) {
+    // TODO(hlundin): Write test for this.
+    if (decoder)
+      decoder->Reset();
+
+    // Reset comfort noise decoder.
+    ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+    if (cng_decoder)
+      cng_decoder->Reset();
+
+    reset_decoder_ = false;
+  }
+
+  *decoded_length = 0;
+  // Update codec-internal PLC state.
+  if ((*operation == Operation::kMerge) && decoder && decoder->HasDecodePlc()) {
+    decoder->DecodePlc(1, &decoded_buffer_[*decoded_length]);
+  }
+
+  int return_value;
+  if (*operation == Operation::kCodecInternalCng) {
+    RTC_DCHECK(packet_list->empty());
+    return_value = DecodeCng(decoder, decoded_length, speech_type);
+  } else {
+    return_value = DecodeLoop(packet_list, *operation, decoder, decoded_length,
+                              speech_type);
+  }
+
+  if (*decoded_length < 0) {
+    // Error returned from the decoder.
+    *decoded_length = 0;
+    sync_buffer_->IncreaseEndTimestamp(
+        static_cast<uint32_t>(decoder_frame_length_));
+    int error_code = 0;
+    if (decoder)
+      error_code = decoder->ErrorCode();
+    if (error_code != 0) {
+      // Got some error code from the decoder.
+      return_value = kDecoderErrorCode;
+      RTC_LOG(LS_WARNING) << "Decoder returned error code: " << error_code;
+    } else {
+      // Decoder does not implement error codes. Return generic error.
+      return_value = kOtherDecoderError;
+      RTC_LOG(LS_WARNING) << "Decoder error (no error code)";
+    }
+    *operation = Operation::kExpand;  // Do expansion to get data instead.
+  }
+  if (*speech_type != AudioDecoder::kComfortNoise) {
+    // Don't increment timestamp if codec returned CNG speech type
+    // since in this case, the we will increment the CNGplayedTS counter.
+    // Increase with number of samples per channel.
+    RTC_DCHECK(*decoded_length == 0 ||
+               (decoder && decoder->Channels() == sync_buffer_->Channels()));
+    sync_buffer_->IncreaseEndTimestamp(
+        *decoded_length / static_cast<int>(sync_buffer_->Channels()));
+  }
+  return return_value;
+}
+
+int NetEqImpl::DecodeCng(AudioDecoder* decoder,
+                         int* decoded_length,
+                         AudioDecoder::SpeechType* speech_type) {
+  if (!decoder) {
+    // This happens when active decoder is not defined.
+    *decoded_length = -1;
+    return 0;
+  }
+
+  while (*decoded_length < rtc::dchecked_cast<int>(output_size_samples_)) {
+    const int length = decoder->Decode(
+        nullptr, 0, fs_hz_,
+        (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
+        &decoded_buffer_[*decoded_length], speech_type);
+    if (length > 0) {
+      *decoded_length += length;
+    } else {
+      // Error.
+      RTC_LOG(LS_WARNING) << "Failed to decode CNG";
+      *decoded_length = -1;
+      break;
+    }
+    if (*decoded_length > static_cast<int>(decoded_buffer_length_)) {
+      // Guard against overflow.
+      RTC_LOG(LS_WARNING) << "Decoded too much CNG.";
+      return kDecodedTooMuch;
+    }
+  }
+  stats_->GeneratedNoiseSamples(*decoded_length);
+  return 0;
+}
+
+int NetEqImpl::DecodeLoop(PacketList* packet_list,
+                          const Operation& operation,
+                          AudioDecoder* decoder,
+                          int* decoded_length,
+                          AudioDecoder::SpeechType* speech_type) {
+  RTC_DCHECK(last_decoded_packet_infos_.empty());
+
+  // Do decoding.
+  while (!packet_list->empty() && !decoder_database_->IsComfortNoise(
+                                      packet_list->front().payload_type)) {
+    RTC_DCHECK(decoder);  // At this point, we must have a decoder object.
+    // The number of channels in the `sync_buffer_` should be the same as the
+    // number decoder channels.
+    RTC_DCHECK_EQ(sync_buffer_->Channels(), decoder->Channels());
+    RTC_DCHECK_GE(decoded_buffer_length_, kMaxFrameSize * decoder->Channels());
+    RTC_DCHECK(operation == Operation::kNormal ||
+               operation == Operation::kAccelerate ||
+               operation == Operation::kFastAccelerate ||
+               operation == Operation::kMerge ||
+               operation == Operation::kPreemptiveExpand);
+
+    auto opt_result = packet_list->front().frame->Decode(
+        rtc::ArrayView<int16_t>(&decoded_buffer_[*decoded_length],
+                                decoded_buffer_length_ - *decoded_length));
+    last_decoded_packet_infos_.push_back(
+        std::move(packet_list->front().packet_info));
+    packet_list->pop_front();
+    if (opt_result) {
+      const auto& result = *opt_result;
+      *speech_type = result.speech_type;
+      if (result.num_decoded_samples > 0) {
+        *decoded_length += rtc::dchecked_cast<int>(result.num_decoded_samples);
+        // Update `decoder_frame_length_` with number of samples per channel.
+        decoder_frame_length_ =
+            result.num_decoded_samples / decoder->Channels();
+      }
+    } else {
+      // Error.
+      // TODO(ossu): What to put here?
+      RTC_LOG(LS_WARNING) << "Decode error";
+      *decoded_length = -1;
+      last_decoded_packet_infos_.clear();
+      packet_list->clear();
+      break;
+    }
+    if (*decoded_length > rtc::dchecked_cast<int>(decoded_buffer_length_)) {
+      // Guard against overflow.
+      RTC_LOG(LS_WARNING) << "Decoded too much.";
+      packet_list->clear();
+      return kDecodedTooMuch;
+    }
+  }  // End of decode loop.
+
+  // If the list is not empty at this point, either a decoding error terminated
+  // the while-loop, or list must hold exactly one CNG packet.
+  RTC_DCHECK(
+      packet_list->empty() || *decoded_length < 0 ||
+      (packet_list->size() == 1 &&
+       decoder_database_->IsComfortNoise(packet_list->front().payload_type)));
+  return 0;
+}
+
+void NetEqImpl::DoNormal(const int16_t* decoded_buffer,
+                         size_t decoded_length,
+                         AudioDecoder::SpeechType speech_type,
+                         bool play_dtmf) {
+  RTC_DCHECK(normal_.get());
+  normal_->Process(decoded_buffer, decoded_length, last_mode_,
+                   algorithm_buffer_.get());
+  if (decoded_length != 0) {
+    last_mode_ = Mode::kNormal;
+  }
+
+  // If last packet was decoded as an inband CNG, set mode to CNG instead.
+  if ((speech_type == AudioDecoder::kComfortNoise) ||
+      ((last_mode_ == Mode::kCodecInternalCng) && (decoded_length == 0))) {
+    // TODO(hlundin): Remove second part of || statement above.
+    last_mode_ = Mode::kCodecInternalCng;
+  }
+
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+}
+
+void NetEqImpl::DoMerge(int16_t* decoded_buffer,
+                        size_t decoded_length,
+                        AudioDecoder::SpeechType speech_type,
+                        bool play_dtmf) {
+  RTC_DCHECK(merge_.get());
+  size_t new_length =
+      merge_->Process(decoded_buffer, decoded_length, algorithm_buffer_.get());
+  // Correction can be negative.
+  int expand_length_correction =
+      rtc::dchecked_cast<int>(new_length) -
+      rtc::dchecked_cast<int>(decoded_length / algorithm_buffer_->Channels());
+
+  // Update in-call and post-call statistics.
+  if (expand_->MuteFactor(0) == 0) {
+    // Expand generates only noise.
+    stats_->ExpandedNoiseSamplesCorrection(expand_length_correction);
+  } else {
+    // Expansion generates more than only noise.
+    stats_->ExpandedVoiceSamplesCorrection(expand_length_correction);
+  }
+
+  last_mode_ = Mode::kMerge;
+  // If last packet was decoded as an inband CNG, set mode to CNG instead.
+  if (speech_type == AudioDecoder::kComfortNoise) {
+    last_mode_ = Mode::kCodecInternalCng;
+  }
+  expand_->Reset();
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+}
+
+bool NetEqImpl::DoCodecPlc() {
+  AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
+  if (!decoder) {
+    return false;
+  }
+  const size_t channels = algorithm_buffer_->Channels();
+  const size_t requested_samples_per_channel =
+      output_size_samples_ -
+      (sync_buffer_->FutureLength() - expand_->overlap_length());
+  concealment_audio_.Clear();
+  decoder->GeneratePlc(requested_samples_per_channel, &concealment_audio_);
+  if (concealment_audio_.empty()) {
+    // Nothing produced. Resort to regular expand.
+    return false;
+  }
+  RTC_CHECK_GE(concealment_audio_.size(),
+               requested_samples_per_channel * channels);
+  sync_buffer_->PushBackInterleaved(concealment_audio_);
+  RTC_DCHECK_NE(algorithm_buffer_->Channels(), 0);
+  const size_t concealed_samples_per_channel =
+      concealment_audio_.size() / channels;
+
+  // Update in-call and post-call statistics.
+  const bool is_new_concealment_event = (last_mode_ != Mode::kCodecPlc);
+  if (std::all_of(concealment_audio_.cbegin(), concealment_audio_.cend(),
+                  [](int16_t i) { return i == 0; })) {
+    // Expand operation generates only noise.
+    stats_->ExpandedNoiseSamples(concealed_samples_per_channel,
+                                 is_new_concealment_event);
+  } else {
+    // Expand operation generates more than only noise.
+    stats_->ExpandedVoiceSamples(concealed_samples_per_channel,
+                                 is_new_concealment_event);
+  }
+  last_mode_ = Mode::kCodecPlc;
+  if (!generated_noise_stopwatch_) {
+    // Start a new stopwatch since we may be covering for a lost CNG packet.
+    generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch();
+  }
+  return true;
+}
+
+int NetEqImpl::DoExpand(bool play_dtmf) {
+  while ((sync_buffer_->FutureLength() - expand_->overlap_length()) <
+         output_size_samples_) {
+    algorithm_buffer_->Clear();
+    int return_value = expand_->Process(algorithm_buffer_.get());
+    size_t length = algorithm_buffer_->Size();
+    bool is_new_concealment_event = (last_mode_ != Mode::kExpand);
+
+    // Update in-call and post-call statistics.
+    if (expand_->MuteFactor(0) == 0) {
+      // Expand operation generates only noise.
+      stats_->ExpandedNoiseSamples(length, is_new_concealment_event);
+    } else {
+      // Expand operation generates more than only noise.
+      stats_->ExpandedVoiceSamples(length, is_new_concealment_event);
+    }
+
+    last_mode_ = Mode::kExpand;
+
+    if (return_value < 0) {
+      return return_value;
+    }
+
+    sync_buffer_->PushBack(*algorithm_buffer_);
+    algorithm_buffer_->Clear();
+  }
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+
+  if (!generated_noise_stopwatch_) {
+    // Start a new stopwatch since we may be covering for a lost CNG packet.
+    generated_noise_stopwatch_ = tick_timer_->GetNewStopwatch();
+  }
+
+  return 0;
+}
+
+int NetEqImpl::DoAccelerate(int16_t* decoded_buffer,
+                            size_t decoded_length,
+                            AudioDecoder::SpeechType speech_type,
+                            bool play_dtmf,
+                            bool fast_accelerate) {
+  const size_t required_samples =
+      static_cast<size_t>(240 * fs_mult_);  // Must have 30 ms.
+  size_t borrowed_samples_per_channel = 0;
+  size_t num_channels = algorithm_buffer_->Channels();
+  size_t decoded_length_per_channel = decoded_length / num_channels;
+  if (decoded_length_per_channel < required_samples) {
+    // Must move data from the `sync_buffer_` in order to get 30 ms.
+    borrowed_samples_per_channel =
+        static_cast<int>(required_samples - decoded_length_per_channel);
+    memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
+            decoded_buffer, sizeof(int16_t) * decoded_length);
+    sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel,
+                                         decoded_buffer);
+    decoded_length = required_samples * num_channels;
+  }
+
+  size_t samples_removed = 0;
+  Accelerate::ReturnCodes return_code =
+      accelerate_->Process(decoded_buffer, decoded_length, fast_accelerate,
+                           algorithm_buffer_.get(), &samples_removed);
+  stats_->AcceleratedSamples(samples_removed);
+  switch (return_code) {
+    case Accelerate::kSuccess:
+      last_mode_ = Mode::kAccelerateSuccess;
+      break;
+    case Accelerate::kSuccessLowEnergy:
+      last_mode_ = Mode::kAccelerateLowEnergy;
+      break;
+    case Accelerate::kNoStretch:
+      last_mode_ = Mode::kAccelerateFail;
+      break;
+    case Accelerate::kError:
+      // TODO(hlundin): Map to Modes::kError instead?
+      last_mode_ = Mode::kAccelerateFail;
+      return kAccelerateError;
+  }
+
+  if (borrowed_samples_per_channel > 0) {
+    // Copy borrowed samples back to the `sync_buffer_`.
+    size_t length = algorithm_buffer_->Size();
+    if (length < borrowed_samples_per_channel) {
+      // This destroys the beginning of the buffer, but will not cause any
+      // problems.
+      sync_buffer_->ReplaceAtIndex(
+          *algorithm_buffer_,
+          sync_buffer_->Size() - borrowed_samples_per_channel);
+      sync_buffer_->PushFrontZeros(borrowed_samples_per_channel - length);
+      algorithm_buffer_->PopFront(length);
+      RTC_DCHECK(algorithm_buffer_->Empty());
+    } else {
+      sync_buffer_->ReplaceAtIndex(
+          *algorithm_buffer_, borrowed_samples_per_channel,
+          sync_buffer_->Size() - borrowed_samples_per_channel);
+      algorithm_buffer_->PopFront(borrowed_samples_per_channel);
+    }
+  }
+
+  // If last packet was decoded as an inband CNG, set mode to CNG instead.
+  if (speech_type == AudioDecoder::kComfortNoise) {
+    last_mode_ = Mode::kCodecInternalCng;
+  }
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+  expand_->Reset();
+  return 0;
+}
+
+int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
+                                  size_t decoded_length,
+                                  AudioDecoder::SpeechType speech_type,
+                                  bool play_dtmf) {
+  const size_t required_samples =
+      static_cast<size_t>(240 * fs_mult_);  // Must have 30 ms.
+  size_t num_channels = algorithm_buffer_->Channels();
+  size_t borrowed_samples_per_channel = 0;
+  size_t old_borrowed_samples_per_channel = 0;
+  size_t decoded_length_per_channel = decoded_length / num_channels;
+  if (decoded_length_per_channel < required_samples) {
+    // Must move data from the `sync_buffer_` in order to get 30 ms.
+    borrowed_samples_per_channel =
+        required_samples - decoded_length_per_channel;
+    // Calculate how many of these were already played out.
+    old_borrowed_samples_per_channel =
+        (borrowed_samples_per_channel > sync_buffer_->FutureLength())
+            ? (borrowed_samples_per_channel - sync_buffer_->FutureLength())
+            : 0;
+    memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
+            decoded_buffer, sizeof(int16_t) * decoded_length);
+    sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel,
+                                         decoded_buffer);
+    decoded_length = required_samples * num_channels;
+  }
+
+  size_t samples_added = 0;
+  PreemptiveExpand::ReturnCodes return_code = preemptive_expand_->Process(
+      decoded_buffer, decoded_length, old_borrowed_samples_per_channel,
+      algorithm_buffer_.get(), &samples_added);
+  stats_->PreemptiveExpandedSamples(samples_added);
+  switch (return_code) {
+    case PreemptiveExpand::kSuccess:
+      last_mode_ = Mode::kPreemptiveExpandSuccess;
+      break;
+    case PreemptiveExpand::kSuccessLowEnergy:
+      last_mode_ = Mode::kPreemptiveExpandLowEnergy;
+      break;
+    case PreemptiveExpand::kNoStretch:
+      last_mode_ = Mode::kPreemptiveExpandFail;
+      break;
+    case PreemptiveExpand::kError:
+      // TODO(hlundin): Map to Modes::kError instead?
+      last_mode_ = Mode::kPreemptiveExpandFail;
+      return kPreemptiveExpandError;
+  }
+
+  if (borrowed_samples_per_channel > 0) {
+    // Copy borrowed samples back to the `sync_buffer_`.
+    sync_buffer_->ReplaceAtIndex(
+        *algorithm_buffer_, borrowed_samples_per_channel,
+        sync_buffer_->Size() - borrowed_samples_per_channel);
+    algorithm_buffer_->PopFront(borrowed_samples_per_channel);
+  }
+
+  // If last packet was decoded as an inband CNG, set mode to CNG instead.
+  if (speech_type == AudioDecoder::kComfortNoise) {
+    last_mode_ = Mode::kCodecInternalCng;
+  }
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+  expand_->Reset();
+  return 0;
+}
+
+int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
+  if (!packet_list->empty()) {
+    // Must have exactly one SID frame at this point.
+    RTC_DCHECK_EQ(packet_list->size(), 1);
+    const Packet& packet = packet_list->front();
+    if (!decoder_database_->IsComfortNoise(packet.payload_type)) {
+      RTC_LOG(LS_ERROR) << "Trying to decode non-CNG payload as CNG.";
+      return kOtherError;
+    }
+    if (comfort_noise_->UpdateParameters(packet) ==
+        ComfortNoise::kInternalError) {
+      algorithm_buffer_->Zeros(output_size_samples_);
+      return -comfort_noise_->internal_error_code();
+    }
+  }
+  int cn_return =
+      comfort_noise_->Generate(output_size_samples_, algorithm_buffer_.get());
+  expand_->Reset();
+  last_mode_ = Mode::kRfc3389Cng;
+  if (!play_dtmf) {
+    dtmf_tone_generator_->Reset();
+  }
+  if (cn_return == ComfortNoise::kInternalError) {
+    RTC_LOG(LS_WARNING) << "Comfort noise generator returned error code: "
+                        << comfort_noise_->internal_error_code();
+    return kComfortNoiseErrorCode;
+  } else if (cn_return == ComfortNoise::kUnknownPayloadType) {
+    return kUnknownRtpPayloadType;
+  }
+  return 0;
+}
+
+void NetEqImpl::DoCodecInternalCng(const int16_t* decoded_buffer,
+                                   size_t decoded_length) {
+  RTC_DCHECK(normal_.get());
+  normal_->Process(decoded_buffer, decoded_length, last_mode_,
+                   algorithm_buffer_.get());
+  last_mode_ = Mode::kCodecInternalCng;
+  expand_->Reset();
+}
+
+int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
+  // This block of the code and the block further down, handling `dtmf_switch`
+  // are commented out. Otherwise playing out-of-band DTMF would fail in VoE
+  // test, DtmfTest.ManualSuccessfullySendsOutOfBandTelephoneEvents. This is
+  // equivalent to `dtmf_switch` always be false.
+  //
+  // See http://webrtc-codereview.appspot.com/1195004/ for discussion
+  // On this issue. This change might cause some glitches at the point of
+  // switch from audio to DTMF. Issue 1545 is filed to track this.
+  //
+  //  bool dtmf_switch = false;
+  //  if ((last_mode_ != Modes::kDtmf) &&
+  //      dtmf_tone_generator_->initialized()) {
+  //    // Special case; see below.
+  //    // We must catch this before calling Generate, since `initialized` is
+  //    // modified in that call.
+  //    dtmf_switch = true;
+  //  }
+
+  int dtmf_return_value = 0;
+  if (!dtmf_tone_generator_->initialized()) {
+    // Initialize if not already done.
+    dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no,
+                                                   dtmf_event.volume);
+  }
+
+  if (dtmf_return_value == 0) {
+    // Generate DTMF signal.
+    dtmf_return_value = dtmf_tone_generator_->Generate(output_size_samples_,
+                                                       algorithm_buffer_.get());
+  }
+
+  if (dtmf_return_value < 0) {
+    algorithm_buffer_->Zeros(output_size_samples_);
+    return dtmf_return_value;
+  }
+
+  //  if (dtmf_switch) {
+  //    // This is the special case where the previous operation was DTMF
+  //    // overdub, but the current instruction is "regular" DTMF. We must make
+  //    // sure that the DTMF does not have any discontinuities. The first DTMF
+  //    // sample that we generate now must be played out immediately, therefore
+  //    // it must be copied to the speech buffer.
+  //    // TODO(hlundin): This code seems incorrect. (Legacy.) Write test and
+  //    // verify correct operation.
+  //    RTC_DCHECK_NOTREACHED();
+  //    // Must generate enough data to replace all of the `sync_buffer_`
+  //    // "future".
+  //    int required_length = sync_buffer_->FutureLength();
+  //    RTC_DCHECK(dtmf_tone_generator_->initialized());
+  //    dtmf_return_value = dtmf_tone_generator_->Generate(required_length,
+  //                                                       algorithm_buffer_);
+  //    RTC_DCHECK((size_t) required_length == algorithm_buffer_->Size());
+  //    if (dtmf_return_value < 0) {
+  //      algorithm_buffer_->Zeros(output_size_samples_);
+  //      return dtmf_return_value;
+  //    }
+  //
+  //    // Overwrite the "future" part of the speech buffer with the new DTMF
+  //    // data.
+  //    // TODO(hlundin): It seems that this overwriting has gone lost.
+  //    // Not adapted for multi-channel yet.
+  //    RTC_DCHECK(algorithm_buffer_->Channels() == 1);
+  //    if (algorithm_buffer_->Channels() != 1) {
+  //      RTC_LOG(LS_WARNING) << "DTMF not supported for more than one channel";
+  //      return kStereoNotSupported;
+  //    }
+  //    // Shuffle the remaining data to the beginning of algorithm buffer.
+  //    algorithm_buffer_->PopFront(sync_buffer_->FutureLength());
+  //  }
+
+  sync_buffer_->IncreaseEndTimestamp(
+      static_cast<uint32_t>(output_size_samples_));
+  expand_->Reset();
+  last_mode_ = Mode::kDtmf;
+
+  // Set to false because the DTMF is already in the algorithm buffer.
+  *play_dtmf = false;
+  return 0;
+}
+
+int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event,
+                           size_t num_channels,
+                           int16_t* output) const {
+  size_t out_index = 0;
+  size_t overdub_length = output_size_samples_;  // Default value.
+
+  if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) {
+    // Special operation for transition from "DTMF only" to "DTMF overdub".
+    out_index =
+        std::min(sync_buffer_->dtmf_index() - sync_buffer_->next_index(),
+                 output_size_samples_);
+    overdub_length = output_size_samples_ - out_index;
+  }
+
+  AudioMultiVector dtmf_output(num_channels);
+  int dtmf_return_value = 0;
+  if (!dtmf_tone_generator_->initialized()) {
+    dtmf_return_value = dtmf_tone_generator_->Init(fs_hz_, dtmf_event.event_no,
+                                                   dtmf_event.volume);
+  }
+  if (dtmf_return_value == 0) {
+    dtmf_return_value =
+        dtmf_tone_generator_->Generate(overdub_length, &dtmf_output);
+    RTC_DCHECK_EQ(overdub_length, dtmf_output.Size());
+  }
+  dtmf_output.ReadInterleaved(overdub_length, &output[out_index]);
+  return dtmf_return_value < 0 ? dtmf_return_value : 0;
+}
+
+int NetEqImpl::ExtractPackets(size_t required_samples,
+                              PacketList* packet_list) {
+  bool first_packet = true;
+  uint8_t prev_payload_type = 0;
+  uint32_t prev_timestamp = 0;
+  uint16_t prev_sequence_number = 0;
+  bool next_packet_available = false;
+
+  const Packet* next_packet = packet_buffer_->PeekNextPacket();
+  RTC_DCHECK(next_packet);
+  if (!next_packet) {
+    RTC_LOG(LS_ERROR) << "Packet buffer unexpectedly empty.";
+    return -1;
+  }
+  uint32_t first_timestamp = next_packet->timestamp;
+  size_t extracted_samples = 0;
+
+  // Packet extraction loop.
+  do {
+    timestamp_ = next_packet->timestamp;
+    absl::optional<Packet> packet = packet_buffer_->GetNextPacket();
+    // `next_packet` may be invalid after the `packet_buffer_` operation.
+    next_packet = nullptr;
+    if (!packet) {
+      RTC_LOG(LS_ERROR) << "Should always be able to extract a packet here";
+      RTC_DCHECK_NOTREACHED();  // Should always be able to extract a packet
+                                // here.
+      return -1;
+    }
+    const uint64_t waiting_time_ms = packet->waiting_time->ElapsedMs();
+    stats_->StoreWaitingTime(waiting_time_ms);
+    RTC_DCHECK(!packet->empty());
+
+    if (first_packet) {
+      first_packet = false;
+      if (nack_enabled_) {
+        RTC_DCHECK(nack_);
+        // TODO(henrik.lundin): Should we update this for all decoded packets?
+        nack_->UpdateLastDecodedPacket(packet->sequence_number,
+                                       packet->timestamp);
+      }
+      prev_sequence_number = packet->sequence_number;
+      prev_timestamp = packet->timestamp;
+      prev_payload_type = packet->payload_type;
+    }
+
+    const bool has_cng_packet =
+        decoder_database_->IsComfortNoise(packet->payload_type);
+    // Store number of extracted samples.
+    size_t packet_duration = 0;
+    if (packet->frame) {
+      packet_duration = packet->frame->Duration();
+      // TODO(ossu): Is this the correct way to track Opus FEC packets?
+      if (packet->priority.codec_level > 0) {
+        stats_->SecondaryDecodedSamples(
+            rtc::dchecked_cast<int>(packet_duration));
+      }
+    } else if (!has_cng_packet) {
+      RTC_LOG(LS_WARNING) << "Unknown payload type "
+                          << static_cast<int>(packet->payload_type);
+      RTC_DCHECK_NOTREACHED();
+    }
+
+    if (packet_duration == 0) {
+      // Decoder did not return a packet duration. Assume that the packet
+      // contains the same number of samples as the previous one.
+      packet_duration = decoder_frame_length_;
+    }
+    extracted_samples = packet->timestamp - first_timestamp + packet_duration;
+
+    RTC_DCHECK(controller_);
+    stats_->JitterBufferDelay(packet_duration, waiting_time_ms,
+                              controller_->TargetLevelMs(),
+                              controller_->UnlimitedTargetLevelMs());
+
+    packet_list->push_back(std::move(*packet));  // Store packet in list.
+    packet = absl::nullopt;  // Ensure it's never used after the move.
+
+    // Check what packet is available next.
+    next_packet = packet_buffer_->PeekNextPacket();
+    next_packet_available = false;
+    if (next_packet && prev_payload_type == next_packet->payload_type &&
+        !has_cng_packet) {
+      int16_t seq_no_diff = next_packet->sequence_number - prev_sequence_number;
+      size_t ts_diff = next_packet->timestamp - prev_timestamp;
+      if ((seq_no_diff == 1 || seq_no_diff == 0) &&
+          ts_diff <= packet_duration) {
+        // The next sequence number is available, or the next part of a packet
+        // that was split into pieces upon insertion.
+        next_packet_available = true;
+      }
+      prev_sequence_number = next_packet->sequence_number;
+      prev_timestamp = next_packet->timestamp;
+    }
+  } while (extracted_samples < required_samples && next_packet_available);
+
+  if (extracted_samples > 0) {
+    // Delete old packets only when we are going to decode something. Otherwise,
+    // we could end up in the situation where we never decode anything, since
+    // all incoming packets are considered too old but the buffer will also
+    // never be flooded and flushed.
+    packet_buffer_->DiscardAllOldPackets(timestamp_, stats_.get());
+  }
+
+  return rtc::dchecked_cast<int>(extracted_samples);
+}
+
+void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) {
+  // Delete objects and create new ones.
+  expand_.reset(expand_factory_->Create(background_noise_.get(),
+                                        sync_buffer_.get(), &random_vector_,
+                                        stats_.get(), fs_hz, channels));
+  merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get()));
+}
+
+void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
+  RTC_LOG(LS_VERBOSE) << "SetSampleRateAndChannels " << fs_hz << " "
+                      << channels;
+  // TODO(hlundin): Change to an enumerator and skip assert.
+  RTC_DCHECK(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 ||
+             fs_hz == 48000);
+  RTC_DCHECK_GT(channels, 0);
+
+  // Before changing the sample rate, end and report any ongoing expand event.
+  stats_->EndExpandEvent(fs_hz_);
+  fs_hz_ = fs_hz;
+  fs_mult_ = fs_hz / 8000;
+  output_size_samples_ = static_cast<size_t>(kOutputSizeMs * 8 * fs_mult_);
+  decoder_frame_length_ = 3 * output_size_samples_;  // Initialize to 30ms.
+
+  last_mode_ = Mode::kNormal;
+
+  ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+  if (cng_decoder)
+    cng_decoder->Reset();
+
+  // Reinit post-decode VAD with new sample rate.
+  RTC_DCHECK(vad_.get());  // Cannot be NULL here.
+  vad_->Init();
+
+  // Delete algorithm buffer and create a new one.
+  algorithm_buffer_.reset(new AudioMultiVector(channels));
+
+  // Delete sync buffer and create a new one.
+  sync_buffer_.reset(new SyncBuffer(channels, kSyncBufferSize * fs_mult_));
+
+  // Delete BackgroundNoise object and create a new one.
+  background_noise_.reset(new BackgroundNoise(channels));
+
+  // Reset random vector.
+  random_vector_.Reset();
+
+  UpdatePlcComponents(fs_hz, channels);
+
+  // Move index so that we create a small set of future samples (all 0).
+  sync_buffer_->set_next_index(sync_buffer_->next_index() -
+                               expand_->overlap_length());
+
+  normal_.reset(new Normal(fs_hz, decoder_database_.get(), *background_noise_,
+                           expand_.get(), stats_.get()));
+  accelerate_.reset(
+      accelerate_factory_->Create(fs_hz, channels, *background_noise_));
+  preemptive_expand_.reset(preemptive_expand_factory_->Create(
+      fs_hz, channels, *background_noise_, expand_->overlap_length()));
+
+  // Delete ComfortNoise object and create a new one.
+  comfort_noise_.reset(
+      new ComfortNoise(fs_hz, decoder_database_.get(), sync_buffer_.get()));
+
+  // Verify that `decoded_buffer_` is long enough.
+  if (decoded_buffer_length_ < kMaxFrameSize * channels) {
+    // Reallocate to larger size.
+    decoded_buffer_length_ = kMaxFrameSize * channels;
+    decoded_buffer_.reset(new int16_t[decoded_buffer_length_]);
+  }
+  RTC_CHECK(controller_) << "Unexpectedly found no NetEqController";
+  controller_->SetSampleRate(fs_hz_, output_size_samples_);
+}
+
+NetEqImpl::OutputType NetEqImpl::LastOutputType() {
+  RTC_DCHECK(vad_.get());
+  RTC_DCHECK(expand_.get());
+  if (last_mode_ == Mode::kCodecInternalCng ||
+      last_mode_ == Mode::kRfc3389Cng) {
+    return OutputType::kCNG;
+  } else if (last_mode_ == Mode::kExpand && expand_->MuteFactor(0) == 0) {
+    // Expand mode has faded down to background noise only (very long expand).
+    return OutputType::kPLCCNG;
+  } else if (last_mode_ == Mode::kExpand) {
+    return OutputType::kPLC;
+  } else if (vad_->running() && !vad_->active_speech()) {
+    return OutputType::kVadPassive;
+  } else if (last_mode_ == Mode::kCodecPlc) {
+    return OutputType::kCodecPLC;
+  } else {
+    return OutputType::kNormalSpeech;
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h
new file mode 100644
index 0000000000..6120eab5b6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -0,0 +1,404 @@
+/*
+ *  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_NETEQ_NETEQ_IMPL_H_
+#define MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio/audio_frame.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "api/neteq/neteq_controller_factory.h"
+#include "api/neteq/tick_timer.h"
+#include "api/rtp_packet_info.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/expand_uma_logger.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class Accelerate;
+class BackgroundNoise;
+class Clock;
+class ComfortNoise;
+class DecoderDatabase;
+class DtmfBuffer;
+class DtmfToneGenerator;
+class Expand;
+class Merge;
+class NackTracker;
+class Normal;
+class PacketBuffer;
+class RedPayloadSplitter;
+class PostDecodeVad;
+class PreemptiveExpand;
+class RandomVector;
+class SyncBuffer;
+class TimestampScaler;
+struct AccelerateFactory;
+struct DtmfEvent;
+struct ExpandFactory;
+struct PreemptiveExpandFactory;
+
+class NetEqImpl : public webrtc::NetEq {
+ public:
+  enum class OutputType {
+    kNormalSpeech,
+    kPLC,
+    kCNG,
+    kPLCCNG,
+    kVadPassive,
+    kCodecPLC
+  };
+
+  enum ErrorCodes {
+    kNoError = 0,
+    kOtherError,
+    kUnknownRtpPayloadType,
+    kDecoderNotFound,
+    kInvalidPointer,
+    kAccelerateError,
+    kPreemptiveExpandError,
+    kComfortNoiseErrorCode,
+    kDecoderErrorCode,
+    kOtherDecoderError,
+    kInvalidOperation,
+    kDtmfParsingError,
+    kDtmfInsertError,
+    kSampleUnderrun,
+    kDecodedTooMuch,
+    kRedundancySplitError,
+    kPacketBufferCorruption
+  };
+
+  struct Dependencies {
+    // The constructor populates the Dependencies struct with the default
+    // implementations of the objects. They can all be replaced by the user
+    // before sending the struct to the NetEqImpl constructor. However, there
+    // are dependencies between some of the classes inside the struct, so
+    // swapping out one may make it necessary to re-create another one.
+    Dependencies(const NetEq::Config& config,
+                 Clock* clock,
+                 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
+                 const NetEqControllerFactory& controller_factory);
+    ~Dependencies();
+
+    Clock* const clock;
+    std::unique_ptr<TickTimer> tick_timer;
+    std::unique_ptr<StatisticsCalculator> stats;
+    std::unique_ptr<DecoderDatabase> decoder_database;
+    std::unique_ptr<DtmfBuffer> dtmf_buffer;
+    std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
+    std::unique_ptr<PacketBuffer> packet_buffer;
+    std::unique_ptr<NetEqController> neteq_controller;
+    std::unique_ptr<RedPayloadSplitter> red_payload_splitter;
+    std::unique_ptr<TimestampScaler> timestamp_scaler;
+    std::unique_ptr<AccelerateFactory> accelerate_factory;
+    std::unique_ptr<ExpandFactory> expand_factory;
+    std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
+  };
+
+  // Creates a new NetEqImpl object.
+  NetEqImpl(const NetEq::Config& config,
+            Dependencies&& deps,
+            bool create_components = true);
+
+  ~NetEqImpl() override;
+
+  NetEqImpl(const NetEqImpl&) = delete;
+  NetEqImpl& operator=(const NetEqImpl&) = delete;
+
+  // Inserts a new packet into NetEq. Returns 0 on success, -1 on failure.
+  int InsertPacket(const RTPHeader& rtp_header,
+                   rtc::ArrayView<const uint8_t> payload) override;
+
+  void InsertEmptyPacket(const RTPHeader& rtp_header) override;
+
+  int GetAudio(
+      AudioFrame* audio_frame,
+      bool* muted,
+      int* current_sample_rate_hz = nullptr,
+      absl::optional<Operation> action_override = absl::nullopt) override;
+
+  void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
+
+  bool RegisterPayloadType(int rtp_payload_type,
+                           const SdpAudioFormat& audio_format) override;
+
+  // Removes `rtp_payload_type` from the codec database. Returns 0 on success,
+  // -1 on failure.
+  int RemovePayloadType(uint8_t rtp_payload_type) override;
+
+  void RemoveAllPayloadTypes() override;
+
+  bool SetMinimumDelay(int delay_ms) override;
+
+  bool SetMaximumDelay(int delay_ms) override;
+
+  bool SetBaseMinimumDelayMs(int delay_ms) override;
+
+  int GetBaseMinimumDelayMs() const override;
+
+  int TargetDelayMs() const override;
+
+  int FilteredCurrentDelayMs() const override;
+
+  // Writes the current network statistics to `stats`. The statistics are reset
+  // after the call.
+  int NetworkStatistics(NetEqNetworkStatistics* stats) override;
+
+  NetEqNetworkStatistics CurrentNetworkStatistics() const override;
+
+  NetEqLifetimeStatistics GetLifetimeStatistics() const override;
+
+  NetEqOperationsAndState GetOperationsAndState() const override;
+
+  // Enables post-decode VAD. When enabled, GetAudio() will return
+  // kOutputVADPassive when the signal contains no speech.
+  void EnableVad() override;
+
+  // Disables post-decode VAD.
+  void DisableVad() override;
+
+  absl::optional<uint32_t> GetPlayoutTimestamp() const override;
+
+  int last_output_sample_rate_hz() const override;
+
+  absl::optional<DecoderFormat> GetDecoderFormat(
+      int payload_type) const override;
+
+  // Flushes both the packet buffer and the sync buffer.
+  void FlushBuffers() override;
+
+  void EnableNack(size_t max_nack_list_size) override;
+
+  void DisableNack() override;
+
+  std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
+
+  int SyncBufferSizeMs() const override;
+
+  // This accessor method is only intended for testing purposes.
+  const SyncBuffer* sync_buffer_for_test() const;
+  Operation last_operation_for_test() const;
+
+ protected:
+  static const int kOutputSizeMs = 10;
+  static const size_t kMaxFrameSize = 5760;  // 120 ms @ 48 kHz.
+  // TODO(hlundin): Provide a better value for kSyncBufferSize.
+  // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
+  // calculating correlations of current frame against history.
+  static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
+
+  // Inserts a new packet into NetEq. This is used by the InsertPacket method
+  // above. Returns 0 on success, otherwise an error code.
+  // TODO(hlundin): Merge this with InsertPacket above?
+  int InsertPacketInternal(const RTPHeader& rtp_header,
+                           rtc::ArrayView<const uint8_t> payload)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Delivers 10 ms of audio data. The data is written to `audio_frame`.
+  // Returns 0 on success, otherwise an error code.
+  int GetAudioInternal(AudioFrame* audio_frame,
+                       bool* muted,
+                       absl::optional<Operation> action_override)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Provides a decision to the GetAudioInternal method. The decision what to
+  // do is written to `operation`. Packets to decode are written to
+  // `packet_list`, and a DTMF event to play is written to `dtmf_event`. When
+  // DTMF should be played, `play_dtmf` is set to true by the method.
+  // Returns 0 on success, otherwise an error code.
+  int GetDecision(Operation* operation,
+                  PacketList* packet_list,
+                  DtmfEvent* dtmf_event,
+                  bool* play_dtmf,
+                  absl::optional<Operation> action_override)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Decodes the speech packets in `packet_list`, and writes the results to
+  // `decoded_buffer`, which is allocated to hold `decoded_buffer_length`
+  // elements. The length of the decoded data is written to `decoded_length`.
+  // The speech type -- speech or (codec-internal) comfort noise -- is written
+  // to `speech_type`. If `packet_list` contains any SID frames for RFC 3389
+  // comfort noise, those are not decoded.
+  int Decode(PacketList* packet_list,
+             Operation* operation,
+             int* decoded_length,
+             AudioDecoder::SpeechType* speech_type)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method to Decode(). Performs codec internal CNG.
+  int DecodeCng(AudioDecoder* decoder,
+                int* decoded_length,
+                AudioDecoder::SpeechType* speech_type)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method to Decode(). Performs the actual decoding.
+  int DecodeLoop(PacketList* packet_list,
+                 const Operation& operation,
+                 AudioDecoder* decoder,
+                 int* decoded_length,
+                 AudioDecoder::SpeechType* speech_type)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the Normal class to perform the normal operation.
+  void DoNormal(const int16_t* decoded_buffer,
+                size_t decoded_length,
+                AudioDecoder::SpeechType speech_type,
+                bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the Merge class to perform the merge operation.
+  void DoMerge(int16_t* decoded_buffer,
+               size_t decoded_length,
+               AudioDecoder::SpeechType speech_type,
+               bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  bool DoCodecPlc() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the Expand class to perform the expand operation.
+  int DoExpand(bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the Accelerate class to perform the accelerate
+  // operation.
+  int DoAccelerate(int16_t* decoded_buffer,
+                   size_t decoded_length,
+                   AudioDecoder::SpeechType speech_type,
+                   bool play_dtmf,
+                   bool fast_accelerate) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the PreemptiveExpand class to perform the
+  // preemtive expand operation.
+  int DoPreemptiveExpand(int16_t* decoded_buffer,
+                         size_t decoded_length,
+                         AudioDecoder::SpeechType speech_type,
+                         bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
+  // noise. `packet_list` can either contain one SID frame to update the
+  // noise parameters, or no payload at all, in which case the previously
+  // received parameters are used.
+  int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Calls the audio decoder to generate codec-internal comfort noise when
+  // no packet was received.
+  void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Calls the DtmfToneGenerator class to generate DTMF tones.
+  int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Overdub DTMF on top of `output`.
+  int DtmfOverdub(const DtmfEvent& dtmf_event,
+                  size_t num_channels,
+                  int16_t* output) const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Extracts packets from `packet_buffer_` to produce at least
+  // `required_samples` samples. The packets are inserted into `packet_list`.
+  // Returns the number of samples that the packets in the list will produce, or
+  // -1 in case of an error.
+  int ExtractPackets(size_t required_samples, PacketList* packet_list)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Resets various variables and objects to new values based on the sample rate
+  // `fs_hz` and `channels` number audio channels.
+  void SetSampleRateAndChannels(int fs_hz, size_t channels)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Returns the output type for the audio produced by the latest call to
+  // GetAudio().
+  OutputType LastOutputType() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Updates Expand and Merge.
+  virtual void UpdatePlcComponents(int fs_hz, size_t channels)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  NetEqNetworkStatistics CurrentNetworkStatisticsInternal() const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  Clock* const clock_;
+
+  mutable Mutex mutex_;
+  const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<DecoderDatabase> decoder_database_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<DtmfBuffer> dtmf_buffer_ RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<PacketBuffer> packet_buffer_ RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<TimestampScaler> timestamp_scaler_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<ExpandFactory> expand_factory_ RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<AccelerateFactory> accelerate_factory_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
+      RTC_GUARDED_BY(mutex_);
+  const std::unique_ptr<StatisticsCalculator> stats_ RTC_GUARDED_BY(mutex_);
+
+  std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<NetEqController> controller_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<AudioMultiVector> algorithm_buffer_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<SyncBuffer> sync_buffer_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<Expand> expand_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<Normal> normal_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<Merge> merge_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<Accelerate> accelerate_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<PreemptiveExpand> preemptive_expand_ RTC_GUARDED_BY(mutex_);
+  RandomVector random_vector_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(mutex_);
+  int fs_hz_ RTC_GUARDED_BY(mutex_);
+  int fs_mult_ RTC_GUARDED_BY(mutex_);
+  int last_output_sample_rate_hz_ RTC_GUARDED_BY(mutex_);
+  size_t output_size_samples_ RTC_GUARDED_BY(mutex_);
+  size_t decoder_frame_length_ RTC_GUARDED_BY(mutex_);
+  Mode last_mode_ RTC_GUARDED_BY(mutex_);
+  Operation last_operation_ RTC_GUARDED_BY(mutex_);
+  size_t decoded_buffer_length_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(mutex_);
+  uint32_t playout_timestamp_ RTC_GUARDED_BY(mutex_);
+  bool new_codec_ RTC_GUARDED_BY(mutex_);
+  uint32_t timestamp_ RTC_GUARDED_BY(mutex_);
+  bool reset_decoder_ RTC_GUARDED_BY(mutex_);
+  absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(mutex_);
+  absl::optional<uint8_t> current_cng_rtp_payload_type_ RTC_GUARDED_BY(mutex_);
+  bool first_packet_ RTC_GUARDED_BY(mutex_);
+  bool enable_fast_accelerate_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(mutex_);
+  bool nack_enabled_ RTC_GUARDED_BY(mutex_);
+  const bool enable_muted_state_ RTC_GUARDED_BY(mutex_);
+  AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(mutex_) =
+      AudioFrame::kVadPassive;
+  std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
+      RTC_GUARDED_BY(mutex_);
+  std::vector<RtpPacketInfo> last_decoded_packet_infos_ RTC_GUARDED_BY(mutex_);
+  ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(mutex_);
+  ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(mutex_);
+  bool no_time_stretching_ RTC_GUARDED_BY(mutex_);  // Only used for test.
+  rtc::BufferT<int16_t> concealment_audio_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
new file mode 100644
index 0000000000..ce2be656ef
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -0,0 +1,1871 @@
+/*
+ *  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/neteq/neteq_impl.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/default_neteq_controller_factory.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_controller.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "modules/audio_coding/neteq/accelerate.h"
+#include "modules/audio_coding/neteq/decision_logic.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/histogram.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/mock/mock_dtmf_buffer.h"
+#include "modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h"
+#include "modules/audio_coding/neteq/mock/mock_neteq_controller.h"
+#include "modules/audio_coding/neteq/mock/mock_packet_buffer.h"
+#include "modules/audio_coding/neteq/mock/mock_red_payload_splitter.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/timestamp_scaler.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/clock.h"
+#include "test/audio_decoder_proxy_factory.h"
+#include "test/function_audio_decoder_factory.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder.h"
+#include "test/mock_audio_decoder_factory.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::DoAll;
+using ::testing::ElementsAre;
+using ::testing::InSequence;
+using ::testing::Invoke;
+using ::testing::IsEmpty;
+using ::testing::IsNull;
+using ::testing::Pointee;
+using ::testing::Return;
+using ::testing::ReturnNull;
+using ::testing::SetArgPointee;
+using ::testing::SetArrayArgument;
+using ::testing::SizeIs;
+using ::testing::WithArg;
+
+namespace webrtc {
+
+// This function is called when inserting a packet list into the mock packet
+// buffer. The purpose is to delete all inserted packets properly, to avoid
+// memory leaks in the test.
+int DeletePacketsAndReturnOk(PacketList* packet_list) {
+  packet_list->clear();
+  return PacketBuffer::kOK;
+}
+
+class NetEqImplTest : public ::testing::Test {
+ protected:
+  NetEqImplTest() : clock_(0) { config_.sample_rate_hz = 8000; }
+
+  void CreateInstance(
+      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+    ASSERT_TRUE(decoder_factory);
+    config_.enable_muted_state = enable_muted_state_;
+    NetEqImpl::Dependencies deps(config_, &clock_, decoder_factory,
+                                 DefaultNetEqControllerFactory());
+
+    // Get a local pointer to NetEq's TickTimer object.
+    tick_timer_ = deps.tick_timer.get();
+
+    if (use_mock_decoder_database_) {
+      std::unique_ptr<MockDecoderDatabase> mock(new MockDecoderDatabase);
+      mock_decoder_database_ = mock.get();
+      EXPECT_CALL(*mock_decoder_database_, GetActiveCngDecoder())
+          .WillOnce(ReturnNull());
+      deps.decoder_database = std::move(mock);
+    }
+    decoder_database_ = deps.decoder_database.get();
+
+    if (use_mock_dtmf_buffer_) {
+      std::unique_ptr<MockDtmfBuffer> mock(
+          new MockDtmfBuffer(config_.sample_rate_hz));
+      mock_dtmf_buffer_ = mock.get();
+      deps.dtmf_buffer = std::move(mock);
+    }
+    dtmf_buffer_ = deps.dtmf_buffer.get();
+
+    if (use_mock_dtmf_tone_generator_) {
+      std::unique_ptr<MockDtmfToneGenerator> mock(new MockDtmfToneGenerator);
+      mock_dtmf_tone_generator_ = mock.get();
+      deps.dtmf_tone_generator = std::move(mock);
+    }
+    dtmf_tone_generator_ = deps.dtmf_tone_generator.get();
+
+    if (use_mock_packet_buffer_) {
+      std::unique_ptr<MockPacketBuffer> mock(
+          new MockPacketBuffer(config_.max_packets_in_buffer, tick_timer_));
+      mock_packet_buffer_ = mock.get();
+      deps.packet_buffer = std::move(mock);
+    }
+    packet_buffer_ = deps.packet_buffer.get();
+
+    if (use_mock_neteq_controller_) {
+      std::unique_ptr<MockNetEqController> mock(new MockNetEqController());
+      mock_neteq_controller_ = mock.get();
+      deps.neteq_controller = std::move(mock);
+    } else {
+      deps.stats = std::make_unique<StatisticsCalculator>();
+      NetEqController::Config controller_config;
+      controller_config.tick_timer = tick_timer_;
+      controller_config.base_min_delay_ms = config_.min_delay_ms;
+      controller_config.allow_time_stretching = true;
+      controller_config.max_packets_in_buffer = config_.max_packets_in_buffer;
+      controller_config.clock = &clock_;
+      deps.neteq_controller =
+          std::make_unique<DecisionLogic>(std::move(controller_config));
+    }
+    neteq_controller_ = deps.neteq_controller.get();
+
+    if (use_mock_payload_splitter_) {
+      std::unique_ptr<MockRedPayloadSplitter> mock(new MockRedPayloadSplitter);
+      mock_payload_splitter_ = mock.get();
+      deps.red_payload_splitter = std::move(mock);
+    }
+    red_payload_splitter_ = deps.red_payload_splitter.get();
+
+    deps.timestamp_scaler = std::unique_ptr<TimestampScaler>(
+        new TimestampScaler(*deps.decoder_database.get()));
+
+    neteq_.reset(new NetEqImpl(config_, std::move(deps)));
+    ASSERT_TRUE(neteq_ != NULL);
+  }
+
+  void CreateInstance() { CreateInstance(CreateBuiltinAudioDecoderFactory()); }
+
+  void UseNoMocks() {
+    ASSERT_TRUE(neteq_ == NULL) << "Must call UseNoMocks before CreateInstance";
+    use_mock_decoder_database_ = false;
+    use_mock_neteq_controller_ = false;
+    use_mock_dtmf_buffer_ = false;
+    use_mock_dtmf_tone_generator_ = false;
+    use_mock_packet_buffer_ = false;
+    use_mock_payload_splitter_ = false;
+  }
+
+  virtual ~NetEqImplTest() {
+    if (use_mock_decoder_database_) {
+      EXPECT_CALL(*mock_decoder_database_, Die()).Times(1);
+    }
+    if (use_mock_neteq_controller_) {
+      EXPECT_CALL(*mock_neteq_controller_, Die()).Times(1);
+    }
+    if (use_mock_dtmf_buffer_) {
+      EXPECT_CALL(*mock_dtmf_buffer_, Die()).Times(1);
+    }
+    if (use_mock_dtmf_tone_generator_) {
+      EXPECT_CALL(*mock_dtmf_tone_generator_, Die()).Times(1);
+    }
+    if (use_mock_packet_buffer_) {
+      EXPECT_CALL(*mock_packet_buffer_, Die()).Times(1);
+    }
+  }
+
+  void TestDtmfPacket(int sample_rate_hz) {
+    const size_t kPayloadLength = 4;
+    const uint8_t kPayloadType = 110;
+    const int kSampleRateHz = 16000;
+    config_.sample_rate_hz = kSampleRateHz;
+    UseNoMocks();
+    CreateInstance();
+    // Event: 2, E bit, Volume: 17, Length: 4336.
+    uint8_t payload[kPayloadLength] = {0x02, 0x80 + 0x11, 0x10, 0xF0};
+    RTPHeader rtp_header;
+    rtp_header.payloadType = kPayloadType;
+    rtp_header.sequenceNumber = 0x1234;
+    rtp_header.timestamp = 0x12345678;
+    rtp_header.ssrc = 0x87654321;
+
+    EXPECT_TRUE(neteq_->RegisterPayloadType(
+        kPayloadType, SdpAudioFormat("telephone-event", sample_rate_hz, 1)));
+
+    // Insert first packet.
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+    // Pull audio once.
+    const size_t kMaxOutputSize =
+        static_cast<size_t>(10 * kSampleRateHz / 1000);
+    AudioFrame output;
+    bool muted;
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    ASSERT_FALSE(muted);
+    ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+    EXPECT_EQ(1u, output.num_channels_);
+    EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+
+    // DTMF packets are immediately consumed by `InsertPacket()` and won't be
+    // returned by `GetAudio()`.
+    EXPECT_THAT(output.packet_infos_, IsEmpty());
+
+    // Verify first 64 samples of actual output.
+    const std::vector<int16_t> kOutput(
+        {0,     0,     0,     0,     0,     0,     0,     0,     0,     0,
+         -1578, -2816, -3460, -3403, -2709, -1594, -363,  671,   1269,  1328,
+         908,   202,   -513,  -964,  -955,  -431,  504,   1617,  2602,  3164,
+         3101,  2364,  1073,  -511,  -2047, -3198, -3721, -3525, -2688, -1440,
+         -99,   1015,  1663,  1744,  1319,  588,   -171,  -680,  -747,  -315,
+         515,   1512,  2378,  2828,  2674,  1877,  568,   -986,  -2446, -3482,
+         -3864, -3516, -2534, -1163});
+    ASSERT_GE(kMaxOutputSize, kOutput.size());
+    EXPECT_TRUE(std::equal(kOutput.begin(), kOutput.end(), output.data()));
+  }
+
+  std::unique_ptr<NetEqImpl> neteq_;
+  NetEq::Config config_;
+  SimulatedClock clock_;
+  TickTimer* tick_timer_ = nullptr;
+  MockDecoderDatabase* mock_decoder_database_ = nullptr;
+  DecoderDatabase* decoder_database_ = nullptr;
+  bool use_mock_decoder_database_ = true;
+  MockNetEqController* mock_neteq_controller_ = nullptr;
+  NetEqController* neteq_controller_ = nullptr;
+  bool use_mock_neteq_controller_ = true;
+  MockDtmfBuffer* mock_dtmf_buffer_ = nullptr;
+  DtmfBuffer* dtmf_buffer_ = nullptr;
+  bool use_mock_dtmf_buffer_ = true;
+  MockDtmfToneGenerator* mock_dtmf_tone_generator_ = nullptr;
+  DtmfToneGenerator* dtmf_tone_generator_ = nullptr;
+  bool use_mock_dtmf_tone_generator_ = true;
+  MockPacketBuffer* mock_packet_buffer_ = nullptr;
+  PacketBuffer* packet_buffer_ = nullptr;
+  bool use_mock_packet_buffer_ = true;
+  MockRedPayloadSplitter* mock_payload_splitter_ = nullptr;
+  RedPayloadSplitter* red_payload_splitter_ = nullptr;
+  bool use_mock_payload_splitter_ = true;
+  bool enable_muted_state_ = false;
+};
+
+// This tests the interface class NetEq.
+// TODO(hlundin): Move to separate file?
+TEST(NetEq, CreateAndDestroy) {
+  NetEq::Config config;
+  SimulatedClock clock(0);
+  auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+  std::unique_ptr<NetEq> neteq =
+      DefaultNetEqFactory().CreateNetEq(config, decoder_factory, &clock);
+}
+
+TEST_F(NetEqImplTest, RegisterPayloadType) {
+  CreateInstance();
+  constexpr int rtp_payload_type = 0;
+  const SdpAudioFormat format("pcmu", 8000, 1);
+  EXPECT_CALL(*mock_decoder_database_,
+              RegisterPayload(rtp_payload_type, format));
+  neteq_->RegisterPayloadType(rtp_payload_type, format);
+}
+
+TEST_F(NetEqImplTest, RemovePayloadType) {
+  CreateInstance();
+  uint8_t rtp_payload_type = 0;
+  EXPECT_CALL(*mock_decoder_database_, Remove(rtp_payload_type))
+      .WillOnce(Return(DecoderDatabase::kDecoderNotFound));
+  // Check that kOK is returned when database returns kDecoderNotFound, because
+  // removing a payload type that was never registered is not an error.
+  EXPECT_EQ(NetEq::kOK, neteq_->RemovePayloadType(rtp_payload_type));
+}
+
+TEST_F(NetEqImplTest, RemoveAllPayloadTypes) {
+  CreateInstance();
+  EXPECT_CALL(*mock_decoder_database_, RemoveAll()).WillOnce(Return());
+  neteq_->RemoveAllPayloadTypes();
+}
+
+TEST_F(NetEqImplTest, InsertPacket) {
+  using ::testing::AllOf;
+  using ::testing::Field;
+  CreateInstance();
+  const size_t kPayloadLength = 100;
+  const uint8_t kPayloadType = 0;
+  const uint16_t kFirstSequenceNumber = 0x1234;
+  const uint32_t kFirstTimestamp = 0x12345678;
+  const uint32_t kSsrc = 0x87654321;
+  uint8_t payload[kPayloadLength] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = kFirstSequenceNumber;
+  rtp_header.timestamp = kFirstTimestamp;
+  rtp_header.ssrc = kSsrc;
+  Packet fake_packet;
+  fake_packet.payload_type = kPayloadType;
+  fake_packet.sequence_number = kFirstSequenceNumber;
+  fake_packet.timestamp = kFirstTimestamp;
+
+  auto mock_decoder_factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _, _))
+      .WillOnce(Invoke([&](const SdpAudioFormat& format,
+                           absl::optional<AudioCodecPairId> codec_pair_id,
+                           std::unique_ptr<AudioDecoder>* dec) {
+        EXPECT_EQ("pcmu", format.name);
+
+        std::unique_ptr<MockAudioDecoder> mock_decoder(new MockAudioDecoder);
+        EXPECT_CALL(*mock_decoder, Channels()).WillRepeatedly(Return(1));
+        EXPECT_CALL(*mock_decoder, SampleRateHz()).WillRepeatedly(Return(8000));
+        EXPECT_CALL(*mock_decoder, Die()).Times(1);  // Called when deleted.
+
+        *dec = std::move(mock_decoder);
+      }));
+  DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                    absl::nullopt, mock_decoder_factory.get());
+
+  // Expectations for decoder database.
+  EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  // Expectations for packet buffer.
+  EXPECT_CALL(*mock_packet_buffer_, Empty())
+      .WillOnce(Return(false));  // Called once after first packet is inserted.
+  EXPECT_CALL(*mock_packet_buffer_, Flush(_)).Times(1);
+  EXPECT_CALL(*mock_packet_buffer_, InsertPacketList(_, _, _, _, _, _, _, _))
+      .Times(2)
+      .WillRepeatedly(DoAll(SetArgPointee<2>(kPayloadType),
+                            WithArg<0>(Invoke(DeletePacketsAndReturnOk))));
+  // SetArgPointee<2>(kPayloadType) means that the third argument (zero-based
+  // index) is a pointer, and the variable pointed to is set to kPayloadType.
+  // Also invoke the function DeletePacketsAndReturnOk to properly delete all
+  // packets in the list (to avoid memory leaks in the test).
+  EXPECT_CALL(*mock_packet_buffer_, PeekNextPacket())
+      .Times(1)
+      .WillOnce(Return(&fake_packet));
+
+  // Expectations for DTMF buffer.
+  EXPECT_CALL(*mock_dtmf_buffer_, Flush()).Times(1);
+
+  // Expectations for delay manager.
+  {
+    // All expectations within this block must be called in this specific order.
+    InSequence sequence;  // Dummy variable.
+    // Expectations when the first packet is inserted.
+    EXPECT_CALL(
+        *mock_neteq_controller_,
+        PacketArrived(
+            /*fs_hz*/ 8000,
+            /*should_update_stats*/ _,
+            /*info*/
+            AllOf(
+                Field(&NetEqController::PacketArrivedInfo::is_cng_or_dtmf,
+                      false),
+                Field(&NetEqController::PacketArrivedInfo::main_sequence_number,
+                      kFirstSequenceNumber),
+                Field(&NetEqController::PacketArrivedInfo::main_timestamp,
+                      kFirstTimestamp))));
+    EXPECT_CALL(
+        *mock_neteq_controller_,
+        PacketArrived(
+            /*fs_hz*/ 8000,
+            /*should_update_stats*/ _,
+            /*info*/
+            AllOf(
+                Field(&NetEqController::PacketArrivedInfo::is_cng_or_dtmf,
+                      false),
+                Field(&NetEqController::PacketArrivedInfo::main_sequence_number,
+                      kFirstSequenceNumber + 1),
+                Field(&NetEqController::PacketArrivedInfo::main_timestamp,
+                      kFirstTimestamp + 160))));
+  }
+
+  // Insert first packet.
+  neteq_->InsertPacket(rtp_header, payload);
+
+  // Insert second packet.
+  rtp_header.timestamp += 160;
+  rtp_header.sequenceNumber += 1;
+  neteq_->InsertPacket(rtp_header, payload);
+}
+
+TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
+  UseNoMocks();
+  CreateInstance();
+
+  const int kPayloadLengthSamples = 80;
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;  // PCM 16-bit.
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+
+  // Insert packets. The buffer should not flush.
+  for (size_t i = 1; i <= config_.max_packets_in_buffer; ++i) {
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+    rtp_header.timestamp += kPayloadLengthSamples;
+    rtp_header.sequenceNumber += 1;
+    EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
+  }
+
+  // Insert one more packet and make sure the buffer got flushed. That is, it
+  // should only hold one single packet.
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
+  const Packet* test_packet = packet_buffer_->PeekNextPacket();
+  EXPECT_EQ(rtp_header.timestamp, test_packet->timestamp);
+  EXPECT_EQ(rtp_header.sequenceNumber, test_packet->sequence_number);
+}
+
+TEST_F(NetEqImplTest, TestDtmfPacketAVT) {
+  TestDtmfPacket(8000);
+}
+
+TEST_F(NetEqImplTest, TestDtmfPacketAVT16kHz) {
+  TestDtmfPacket(16000);
+}
+
+TEST_F(NetEqImplTest, TestDtmfPacketAVT32kHz) {
+  TestDtmfPacket(32000);
+}
+
+TEST_F(NetEqImplTest, TestDtmfPacketAVT48kHz) {
+  TestDtmfPacket(48000);
+}
+
+// This test verifies that timestamps propagate from the incoming packets
+// through to the sync buffer and to the playout timestamp.
+TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+  rtp_header.numCSRCs = 3;
+  rtp_header.arrOfCSRCs[0] = 43;
+  rtp_header.arrOfCSRCs[1] = 65;
+  rtp_header.arrOfCSRCs[2] = 17;
+
+  // This is a dummy decoder that produces as many output samples as the input
+  // has bytes. The output is an increasing series, starting at 1 for the first
+  // sample, and then increasing by 1 for each sample.
+  class CountingSamplesDecoder : public AudioDecoder {
+   public:
+    CountingSamplesDecoder() : next_value_(1) {}
+
+    // Produce as many samples as input bytes (`encoded_len`).
+    int DecodeInternal(const uint8_t* encoded,
+                       size_t encoded_len,
+                       int /* sample_rate_hz */,
+                       int16_t* decoded,
+                       SpeechType* speech_type) override {
+      for (size_t i = 0; i < encoded_len; ++i) {
+        decoded[i] = next_value_++;
+      }
+      *speech_type = kSpeech;
+      return rtc::checked_cast<int>(encoded_len);
+    }
+
+    void Reset() override { next_value_ = 1; }
+
+    int SampleRateHz() const override { return kSampleRateHz; }
+
+    size_t Channels() const override { return 1; }
+
+    uint16_t next_value() const { return next_value_; }
+
+   private:
+    int16_t next_value_;
+  } decoder_;
+
+  auto decoder_factory =
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&decoder_);
+
+  UseNoMocks();
+  CreateInstance(decoder_factory);
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("L16", 8000, 1)));
+
+  // Insert one packet.
+  clock_.AdvanceTimeMilliseconds(123456);
+  Timestamp expected_receive_time = clock_.CurrentTime();
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_FALSE(muted);
+  ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+
+  // Verify `output.packet_infos_`.
+  ASSERT_THAT(output.packet_infos_, SizeIs(1));
+  {
+    const auto& packet_info = output.packet_infos_[0];
+    EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
+    EXPECT_THAT(packet_info.csrcs(), ElementsAre(43, 65, 17));
+    EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
+    EXPECT_FALSE(packet_info.audio_level().has_value());
+    EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
+  }
+
+  // Start with a simple check that the fake decoder is behaving as expected.
+  EXPECT_EQ(kPayloadLengthSamples,
+            static_cast<size_t>(decoder_.next_value() - 1));
+
+  // The value of the last of the output samples is the same as the number of
+  // samples played from the decoded packet. Thus, this number + the RTP
+  // timestamp should match the playout timestamp.
+  // Wrap the expected value in an absl::optional to compare them as such.
+  EXPECT_EQ(
+      absl::optional<uint32_t>(rtp_header.timestamp +
+                               output.data()[output.samples_per_channel_ - 1]),
+      neteq_->GetPlayoutTimestamp());
+
+  // Check the timestamp for the last value in the sync buffer. This should
+  // be one full frame length ahead of the RTP timestamp.
+  const SyncBuffer* sync_buffer = neteq_->sync_buffer_for_test();
+  ASSERT_TRUE(sync_buffer != NULL);
+  EXPECT_EQ(rtp_header.timestamp + kPayloadLengthSamples,
+            sync_buffer->end_timestamp());
+
+  // Check that the number of samples still to play from the sync buffer add
+  // up with what was already played out.
+  EXPECT_EQ(
+      kPayloadLengthSamples - output.data()[output.samples_per_channel_ - 1],
+      sync_buffer->FutureLength());
+}
+
+TEST_F(NetEqImplTest, ReorderedPacket) {
+  UseNoMocks();
+
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&mock_decoder));
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+  rtp_header.extension.hasAudioLevel = true;
+  rtp_header.extension.audioLevel = 42;
+
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateHz));
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, kPayloadLengthBytes))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
+  int16_t dummy_output[kPayloadLengthSamples] = {0};
+  // The below expectation will make the mock decoder write
+  // `kPayloadLengthSamples` zeros to the output array, and mark it as speech.
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
+                                           kSampleRateHz, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+                                          dummy_output + kPayloadLengthSamples),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
+                      Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("L16", 8000, 1)));
+
+  // Insert one packet.
+  clock_.AdvanceTimeMilliseconds(123456);
+  Timestamp expected_receive_time = clock_.CurrentTime();
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+
+  // Verify `output.packet_infos_`.
+  ASSERT_THAT(output.packet_infos_, SizeIs(1));
+  {
+    const auto& packet_info = output.packet_infos_[0];
+    EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
+    EXPECT_THAT(packet_info.csrcs(), IsEmpty());
+    EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
+    EXPECT_EQ(packet_info.audio_level(), rtp_header.extension.audioLevel);
+    EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
+  }
+
+  // Insert two more packets. The first one is out of order, and is already too
+  // old, the second one is the expected next packet.
+  rtp_header.sequenceNumber -= 1;
+  rtp_header.timestamp -= kPayloadLengthSamples;
+  rtp_header.extension.audioLevel = 1;
+  payload[0] = 1;
+  clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  rtp_header.sequenceNumber += 2;
+  rtp_header.timestamp += 2 * kPayloadLengthSamples;
+  rtp_header.extension.audioLevel = 2;
+  payload[0] = 2;
+  clock_.AdvanceTimeMilliseconds(2000);
+  expected_receive_time = clock_.CurrentTime();
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  // Expect only the second packet to be decoded (the one with "2" as the first
+  // payload byte).
+  EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(2), kPayloadLengthBytes,
+                                           kSampleRateHz, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+                                          dummy_output + kPayloadLengthSamples),
+                      SetArgPointee<4>(AudioDecoder::kSpeech),
+                      Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+
+  // Pull audio once.
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+
+  // Now check the packet buffer, and make sure it is empty, since the
+  // out-of-order packet should have been discarded.
+  EXPECT_TRUE(packet_buffer_->Empty());
+
+  // NetEq `packets_discarded` should capture this packet discard.
+  EXPECT_EQ(1u, neteq_->GetLifetimeStatistics().packets_discarded);
+
+  // Verify `output.packet_infos_`. Expect to only see the second packet.
+  ASSERT_THAT(output.packet_infos_, SizeIs(1));
+  {
+    const auto& packet_info = output.packet_infos_[0];
+    EXPECT_EQ(packet_info.ssrc(), rtp_header.ssrc);
+    EXPECT_THAT(packet_info.csrcs(), IsEmpty());
+    EXPECT_EQ(packet_info.rtp_timestamp(), rtp_header.timestamp);
+    EXPECT_EQ(packet_info.audio_level(), rtp_header.extension.audioLevel);
+    EXPECT_EQ(packet_info.receive_time(), expected_receive_time);
+  }
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// This test verifies that NetEq can handle the situation where the first
+// incoming packet is rejected.
+TEST_F(NetEqImplTest, FirstPacketUnknown) {
+  UseNoMocks();
+  CreateInstance();
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  // Insert one packet. Note that we have not registered any payload type, so
+  // this packet will be rejected.
+  EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_header, payload));
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
+  EXPECT_THAT(output.packet_infos_, IsEmpty());
+
+  // Register the payload type.
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+
+  // Insert 10 packets.
+  for (size_t i = 0; i < 10; ++i) {
+    rtp_header.sequenceNumber++;
+    rtp_header.timestamp += kPayloadLengthSamples;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+    EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
+  }
+
+  // Pull audio repeatedly and make sure we get normal output, that is not PLC.
+  for (size_t i = 0; i < 3; ++i) {
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
+    EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+    EXPECT_EQ(1u, output.num_channels_);
+    EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_)
+        << "NetEq did not decode the packets as expected.";
+    EXPECT_THAT(output.packet_infos_, SizeIs(1));
+  }
+}
+
+// This test verifies that audio interruption is not logged for the initial
+// PLC period before the first packet is deocoded.
+// TODO(henrik.lundin) Maybe move this test to neteq_network_stats_unittest.cc.
+// Make the test parametrized, so that we can test with different initial
+// sample rates in NetEq.
+class NetEqImplTestSampleRateParameter
+    : public NetEqImplTest,
+      public testing::WithParamInterface<int> {
+ protected:
+  NetEqImplTestSampleRateParameter()
+      : NetEqImplTest(), initial_sample_rate_hz_(GetParam()) {
+    config_.sample_rate_hz = initial_sample_rate_hz_;
+  }
+
+  const int initial_sample_rate_hz_;
+};
+
+class NetEqImplTestSdpFormatParameter
+    : public NetEqImplTest,
+      public testing::WithParamInterface<SdpAudioFormat> {
+ protected:
+  NetEqImplTestSdpFormatParameter()
+      : NetEqImplTest(), sdp_format_(GetParam()) {}
+  const SdpAudioFormat sdp_format_;
+};
+
+// This test does the following:
+// 0. Set up NetEq with initial sample rate given by test parameter, and a codec
+//    sample rate of 16000.
+// 1. Start calling GetAudio before inserting any encoded audio. The audio
+//    produced will be PLC.
+// 2. Insert a number of encoded audio packets.
+// 3. Keep calling GetAudio and verify that no audio interruption was logged.
+//    Call GetAudio until NetEq runs out of data again; PLC starts.
+// 4. Insert one more packet.
+// 5. Call GetAudio until that packet is decoded and the PLC ends.
+
+TEST_P(NetEqImplTestSampleRateParameter,
+       NoAudioInterruptionLoggedBeforeFirstDecode) {
+  UseNoMocks();
+  CreateInstance();
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kPayloadSampleRateHz = 16000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kPayloadSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  // Register the payload type.
+  EXPECT_TRUE(neteq_->RegisterPayloadType(
+      kPayloadType, SdpAudioFormat("l16", kPayloadSampleRateHz, 1)));
+
+  // Pull audio several times. No packets have been inserted yet.
+  const size_t initial_output_size =
+      static_cast<size_t>(10 * initial_sample_rate_hz_ / 1000);  // 10 ms
+  AudioFrame output;
+  bool muted;
+  for (int i = 0; i < 100; ++i) {
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    EXPECT_EQ(initial_output_size, output.samples_per_channel_);
+    EXPECT_EQ(1u, output.num_channels_);
+    EXPECT_NE(AudioFrame::kNormalSpeech, output.speech_type_);
+    EXPECT_THAT(output.packet_infos_, IsEmpty());
+  }
+
+  // Lambda for inserting packets.
+  auto insert_packet = [&]() {
+    rtp_header.sequenceNumber++;
+    rtp_header.timestamp += kPayloadLengthSamples;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  };
+  // Insert 10 packets.
+  for (size_t i = 0; i < 10; ++i) {
+    insert_packet();
+    EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
+  }
+
+  // Pull audio repeatedly and make sure we get normal output, that is not PLC.
+  constexpr size_t kOutputSize =
+      static_cast<size_t>(10 * kPayloadSampleRateHz / 1000);  // 10 ms
+  for (size_t i = 0; i < 3; ++i) {
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    EXPECT_EQ(kOutputSize, output.samples_per_channel_);
+    EXPECT_EQ(1u, output.num_channels_);
+    EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_)
+        << "NetEq did not decode the packets as expected.";
+    EXPECT_THAT(output.packet_infos_, SizeIs(1));
+  }
+
+  // Verify that no interruption was logged.
+  auto lifetime_stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(0, lifetime_stats.interruption_count);
+
+  // Keep pulling audio data until a new PLC period is started.
+  size_t count_loops = 0;
+  while (output.speech_type_ == AudioFrame::kNormalSpeech) {
+    // Make sure we don't hang the test if we never go to PLC.
+    ASSERT_LT(++count_loops, 100u);
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  }
+
+  // Insert a few packets to avoid postpone decoding after expand.
+  for (size_t i = 0; i < 5; ++i) {
+    insert_packet();
+  }
+
+  // Pull audio until the newly inserted packet is decoded and the PLC ends.
+  while (output.speech_type_ != AudioFrame::kNormalSpeech) {
+    // Make sure we don't hang the test if we never go to PLC.
+    ASSERT_LT(++count_loops, 100u);
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  }
+
+  // Verify that no interruption was logged.
+  lifetime_stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(0, lifetime_stats.interruption_count);
+}
+
+// This test does the following:
+// 0. Set up NetEq with initial sample rate given by test parameter, and a codec
+//    sample rate of 16000.
+// 1. Insert a number of encoded audio packets.
+// 2. Call GetAudio and verify that decoded audio is produced.
+// 3. Keep calling GetAudio until NetEq runs out of data; PLC starts.
+// 4. Keep calling GetAudio until PLC has been produced for at least 150 ms.
+// 5. Insert one more packet.
+// 6. Call GetAudio until that packet is decoded and the PLC ends.
+// 7. Verify that an interruption was logged.
+
+TEST_P(NetEqImplTestSampleRateParameter, AudioInterruptionLogged) {
+  UseNoMocks();
+  CreateInstance();
+
+  const uint8_t kPayloadType = 17;  // Just an arbitrary number.
+  const int kPayloadSampleRateHz = 16000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kPayloadSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  // Register the payload type.
+  EXPECT_TRUE(neteq_->RegisterPayloadType(
+      kPayloadType, SdpAudioFormat("l16", kPayloadSampleRateHz, 1)));
+
+  // Lambda for inserting packets.
+  auto insert_packet = [&]() {
+    rtp_header.sequenceNumber++;
+    rtp_header.timestamp += kPayloadLengthSamples;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  };
+  // Insert 10 packets.
+  for (size_t i = 0; i < 10; ++i) {
+    insert_packet();
+    EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
+  }
+
+  AudioFrame output;
+  bool muted;
+  // Keep pulling audio data until a new PLC period is started.
+  size_t count_loops = 0;
+  do {
+    // Make sure we don't hang the test if we never go to PLC.
+    ASSERT_LT(++count_loops, 100u);
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  } while (output.speech_type_ == AudioFrame::kNormalSpeech);
+
+  // Pull audio 15 times, which produces 150 ms of output audio. This should
+  // all be produced as PLC. The total length of the gap will then be 150 ms
+  // plus an initial fraction of 10 ms at the start and the end of the PLC
+  // period. In total, less than 170 ms.
+  for (size_t i = 0; i < 15; ++i) {
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    EXPECT_NE(AudioFrame::kNormalSpeech, output.speech_type_);
+  }
+
+  // Insert a few packets to avoid postpone decoding after expand.
+  for (size_t i = 0; i < 5; ++i) {
+    insert_packet();
+  }
+
+  // Pull audio until the newly inserted packet is decoded and the PLC ends.
+  while (output.speech_type_ != AudioFrame::kNormalSpeech) {
+    // Make sure we don't hang the test if we never go to PLC.
+    ASSERT_LT(++count_loops, 100u);
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  }
+
+  // Verify that the interruption was logged.
+  auto lifetime_stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(1, lifetime_stats.interruption_count);
+  EXPECT_GT(lifetime_stats.total_interruption_duration_ms, 150);
+  EXPECT_LT(lifetime_stats.total_interruption_duration_ms, 170);
+}
+
+INSTANTIATE_TEST_SUITE_P(SampleRates,
+                         NetEqImplTestSampleRateParameter,
+                         testing::Values(8000, 16000, 32000, 48000));
+
+TEST_P(NetEqImplTestSdpFormatParameter, GetNackListScaledTimestamp) {
+  UseNoMocks();
+  CreateInstance();
+
+  neteq_->EnableNack(128);
+
+  const uint8_t kPayloadType = 17;  // Just an arbitrary number.
+  const int kPayloadSampleRateHz = sdp_format_.clockrate_hz;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kPayloadSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
+  std::vector<uint8_t> payload(kPayloadLengthBytes, 0);
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType, sdp_format_));
+
+  auto insert_packet = [&](bool lost = false) {
+    rtp_header.sequenceNumber++;
+    rtp_header.timestamp += kPayloadLengthSamples;
+    if (!lost)
+      EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  };
+
+  // Insert and decode 10 packets.
+  for (size_t i = 0; i < 10; ++i) {
+    insert_packet();
+  }
+  AudioFrame output;
+  size_t count_loops = 0;
+  do {
+    bool muted;
+    // Make sure we don't hang the test if we never go to PLC.
+    ASSERT_LT(++count_loops, 100u);
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  } while (output.speech_type_ == AudioFrame::kNormalSpeech);
+
+  insert_packet();
+
+  insert_packet(/*lost=*/true);
+
+  // Ensure packet gets marked as missing.
+  for (int i = 0; i < 5; ++i) {
+    insert_packet();
+  }
+
+  // Missing packet recoverable with 5ms RTT.
+  EXPECT_THAT(neteq_->GetNackList(5), Not(IsEmpty()));
+
+  // No packets should have TimeToPlay > 500ms.
+  EXPECT_THAT(neteq_->GetNackList(500), IsEmpty());
+}
+
+INSTANTIATE_TEST_SUITE_P(GetNackList,
+                         NetEqImplTestSdpFormatParameter,
+                         testing::Values(SdpAudioFormat("g722", 8000, 1),
+                                         SdpAudioFormat("opus", 48000, 2)));
+
+// This test verifies that NetEq can handle comfort noise and enters/quits codec
+// internal CNG mode properly.
+TEST_F(NetEqImplTest, CodecInternalCng) {
+  UseNoMocks();
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&mock_decoder));
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateKhz = 48;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(20 * kSampleRateKhz);  // 20 ms.
+  const size_t kPayloadLengthBytes = 10;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  int16_t dummy_output[kPayloadLengthSamples] = {0};
+
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateKhz * 1000));
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, kPayloadLengthBytes))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
+  // Packed duration when asking the decoder for more CNG data (without a new
+  // packet).
+  EXPECT_CALL(mock_decoder, PacketDuration(nullptr, 0))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("opus", 48000, 2)));
+
+  struct Packet {
+    int sequence_number_delta;
+    int timestamp_delta;
+    AudioDecoder::SpeechType decoder_output_type;
+  };
+  std::vector<Packet> packets = {
+      {0, 0, AudioDecoder::kSpeech},
+      {1, kPayloadLengthSamples, AudioDecoder::kComfortNoise},
+      {2, 2 * kPayloadLengthSamples, AudioDecoder::kSpeech},
+      {1, kPayloadLengthSamples, AudioDecoder::kSpeech}};
+
+  for (size_t i = 0; i < packets.size(); ++i) {
+    rtp_header.sequenceNumber += packets[i].sequence_number_delta;
+    rtp_header.timestamp += packets[i].timestamp_delta;
+    payload[0] = i;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+    // Pointee(x) verifies that first byte of the payload equals x, this makes
+    // it possible to verify that the correct payload is fed to Decode().
+    EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(i), kPayloadLengthBytes,
+                                             kSampleRateKhz * 1000, _, _))
+        .WillOnce(DoAll(SetArrayArgument<3>(
+                            dummy_output, dummy_output + kPayloadLengthSamples),
+                        SetArgPointee<4>(packets[i].decoder_output_type),
+                        Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+  }
+
+  // Expect comfort noise to be returned by the decoder.
+  EXPECT_CALL(mock_decoder,
+              DecodeInternal(IsNull(), 0, kSampleRateKhz * 1000, _, _))
+      .WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
+                                          dummy_output + kPayloadLengthSamples),
+                      SetArgPointee<4>(AudioDecoder::kComfortNoise),
+                      Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
+
+  std::vector<AudioFrame::SpeechType> expected_output = {
+      AudioFrame::kNormalSpeech, AudioFrame::kCNG, AudioFrame::kNormalSpeech};
+  size_t output_index = 0;
+
+  int timeout_counter = 0;
+  while (!packet_buffer_->Empty()) {
+    ASSERT_LT(timeout_counter++, 20) << "Test timed out";
+    AudioFrame output;
+    bool muted;
+    EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+    if (output_index + 1 < expected_output.size() &&
+        output.speech_type_ == expected_output[output_index + 1]) {
+      ++output_index;
+    } else {
+      EXPECT_EQ(output.speech_type_, expected_output[output_index]);
+    }
+  }
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+TEST_F(NetEqImplTest, UnsupportedDecoder) {
+  UseNoMocks();
+  ::testing::NiceMock<MockAudioDecoder> decoder;
+
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&decoder));
+  static const size_t kNetEqMaxFrameSize = 5760;  // 120 ms @ 48 kHz.
+  static const size_t kChannels = 2;
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = 1;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  int16_t dummy_output[kPayloadLengthSamples * kChannels] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  const uint8_t kFirstPayloadValue = 1;
+  const uint8_t kSecondPayloadValue = 2;
+
+  EXPECT_CALL(decoder,
+              PacketDuration(Pointee(kFirstPayloadValue), kPayloadLengthBytes))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kNetEqMaxFrameSize + 1)));
+
+  EXPECT_CALL(decoder, DecodeInternal(Pointee(kFirstPayloadValue), _, _, _, _))
+      .Times(0);
+
+  EXPECT_CALL(decoder, DecodeInternal(Pointee(kSecondPayloadValue),
+                                      kPayloadLengthBytes, kSampleRateHz, _, _))
+      .Times(1)
+      .WillOnce(DoAll(
+          SetArrayArgument<3>(dummy_output,
+                              dummy_output + kPayloadLengthSamples * kChannels),
+          SetArgPointee<4>(AudioDecoder::kSpeech),
+          Return(static_cast<int>(kPayloadLengthSamples * kChannels))));
+
+  EXPECT_CALL(decoder,
+              PacketDuration(Pointee(kSecondPayloadValue), kPayloadLengthBytes))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kNetEqMaxFrameSize)));
+
+  EXPECT_CALL(decoder, SampleRateHz()).WillRepeatedly(Return(kSampleRateHz));
+
+  EXPECT_CALL(decoder, Channels()).WillRepeatedly(Return(kChannels));
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("L16", 8000, 1)));
+
+  // Insert one packet.
+  payload[0] = kFirstPayloadValue;  // This will make Decode() fail.
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  // Insert another packet.
+  payload[0] = kSecondPayloadValue;  // This will make Decode() successful.
+  rtp_header.sequenceNumber++;
+  // The second timestamp needs to be at least 30 ms after the first to make
+  // the second packet get decoded.
+  rtp_header.timestamp += 3 * kPayloadLengthSamples;
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  AudioFrame output;
+  bool muted;
+  // First call to GetAudio will try to decode the "faulty" packet.
+  // Expect kFail return value.
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
+  // Output size and number of channels should be correct.
+  const size_t kExpectedOutputSize = 10 * (kSampleRateHz / 1000) * kChannels;
+  EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels);
+  EXPECT_EQ(kChannels, output.num_channels_);
+  EXPECT_THAT(output.packet_infos_, IsEmpty());
+
+  // Second call to GetAudio will decode the packet that is ok. No errors are
+  // expected.
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels);
+  EXPECT_EQ(kChannels, output.num_channels_);
+  EXPECT_THAT(output.packet_infos_, SizeIs(1));
+
+  // Die isn't called through NiceMock (since it's called by the
+  // MockAudioDecoder constructor), so it needs to be mocked explicitly.
+  EXPECT_CALL(decoder, Die());
+}
+
+// This test inserts packets until the buffer is flushed. After that, it asks
+// NetEq for the network statistics. The purpose of the test is to make sure
+// that even though the buffer size increment is negative (which it becomes when
+// the packet causing a flush is inserted), the packet length stored in the
+// decision logic remains valid.
+TEST_F(NetEqImplTest, FloodBufferAndGetNetworkStats) {
+  UseNoMocks();
+  CreateInstance();
+
+  const size_t kPayloadLengthSamples = 80;
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;  // PCM 16-bit.
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+
+  // Insert packets until the buffer flushes.
+  for (size_t i = 0; i <= config_.max_packets_in_buffer; ++i) {
+    EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+    rtp_header.timestamp += rtc::checked_cast<uint32_t>(kPayloadLengthSamples);
+    ++rtp_header.sequenceNumber;
+  }
+  EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
+
+  // Ask for network statistics. This should not crash.
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(NetEq::kOK, neteq_->NetworkStatistics(&stats));
+}
+
+TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
+  UseNoMocks();
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&mock_decoder));
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const size_t kPayloadLengthSamples =
+      static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateHz));
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
+  int16_t dummy_output[kPayloadLengthSamples] = {0};
+  // The below expectation will make the mock decoder write
+  // `kPayloadLengthSamples` - 5 zeros to the output array, and mark it as
+  // speech. That is, the decoded length is 5 samples shorter than the expected.
+  EXPECT_CALL(mock_decoder,
+              DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+      .WillOnce(
+          DoAll(SetArrayArgument<3>(dummy_output,
+                                    dummy_output + kPayloadLengthSamples - 5),
+                SetArgPointee<4>(AudioDecoder::kSpeech),
+                Return(rtc::checked_cast<int>(kPayloadLengthSamples - 5))));
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("L16", 8000, 1)));
+
+  // Insert one packet.
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+
+  EXPECT_EQ(5u, neteq_->sync_buffer_for_test()->FutureLength());
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+  EXPECT_THAT(output.packet_infos_, SizeIs(1));
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// This test checks the behavior of NetEq when audio decoder fails.
+TEST_F(NetEqImplTest, DecodingError) {
+  UseNoMocks();
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&mock_decoder));
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const int kDecoderErrorCode = -97;  // Any negative number.
+
+  // We let decoder return 5 ms each time, and therefore, 2 packets make 10 ms.
+  const size_t kFrameLengthSamples =
+      static_cast<size_t>(5 * kSampleRateHz / 1000);
+
+  const size_t kPayloadLengthBytes = 1;  // This can be arbitrary.
+
+  uint8_t payload[kPayloadLengthBytes] = {0};
+
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateHz));
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kFrameLengthSamples)));
+  EXPECT_CALL(mock_decoder, ErrorCode()).WillOnce(Return(kDecoderErrorCode));
+  EXPECT_CALL(mock_decoder, HasDecodePlc()).WillOnce(Return(false));
+  int16_t dummy_output[kFrameLengthSamples] = {0};
+
+  {
+    InSequence sequence;  // Dummy variable.
+    // Mock decoder works normally the first time.
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .Times(3)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kSpeech),
+                  Return(rtc::checked_cast<int>(kFrameLengthSamples))))
+        .RetiresOnSaturation();
+
+    // Then mock decoder fails. A common reason for failure can be buffer being
+    // too short
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .WillOnce(Return(-1))
+        .RetiresOnSaturation();
+
+    // Mock decoder finally returns to normal.
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .Times(2)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kSpeech),
+                  Return(rtc::checked_cast<int>(kFrameLengthSamples))));
+  }
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("L16", 8000, 1)));
+
+  // Insert packets.
+  for (int i = 0; i < 20; ++i) {
+    rtp_header.sequenceNumber += 1;
+    rtp_header.timestamp += kFrameLengthSamples;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  }
+
+  // Pull audio.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+  EXPECT_THAT(output.packet_infos_, SizeIs(2));  // 5 ms packets vs 10 ms output
+
+  // Pull audio again. Decoder fails.
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  // We are not expecting anything for output.speech_type_, since an error was
+  // returned.
+
+  // Pull audio again, should behave normal.
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+  EXPECT_THAT(output.packet_infos_, SizeIs(2));  // 5 ms packets vs 10 ms output
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// This test checks the behavior of NetEq when audio decoder fails during CNG.
+TEST_F(NetEqImplTest, DecodingErrorDuringInternalCng) {
+  UseNoMocks();
+
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+  CreateInstance(
+      rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&mock_decoder));
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const int kSampleRateHz = 8000;
+  const int kDecoderErrorCode = -97;  // Any negative number.
+
+  // We let decoder return 5 ms each time, and therefore, 2 packets make 10 ms.
+  const size_t kFrameLengthSamples =
+      static_cast<size_t>(5 * kSampleRateHz / 1000);
+
+  const size_t kPayloadLengthBytes = 1;  // This can be arbitrary.
+
+  uint8_t payload[kPayloadLengthBytes] = {0};
+
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
+  EXPECT_CALL(mock_decoder, SampleRateHz())
+      .WillRepeatedly(Return(kSampleRateHz));
+  EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
+  EXPECT_CALL(mock_decoder, PacketDuration(_, _))
+      .WillRepeatedly(Return(rtc::checked_cast<int>(kFrameLengthSamples)));
+  EXPECT_CALL(mock_decoder, ErrorCode()).WillOnce(Return(kDecoderErrorCode));
+  int16_t dummy_output[kFrameLengthSamples] = {0};
+
+  {
+    InSequence sequence;  // Dummy variable.
+    // Mock decoder works normally the first 2 times.
+    EXPECT_CALL(mock_decoder,
+                DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
+        .Times(2)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kComfortNoise),
+                  Return(rtc::checked_cast<int>(kFrameLengthSamples))))
+        .RetiresOnSaturation();
+
+    // Then mock decoder fails. A common reason for failure can be buffer being
+    // too short
+    EXPECT_CALL(mock_decoder, DecodeInternal(nullptr, 0, kSampleRateHz, _, _))
+        .WillOnce(Return(-1))
+        .RetiresOnSaturation();
+
+    // Mock decoder finally returns to normal.
+    EXPECT_CALL(mock_decoder, DecodeInternal(nullptr, 0, kSampleRateHz, _, _))
+        .Times(2)
+        .WillRepeatedly(
+            DoAll(SetArrayArgument<3>(dummy_output,
+                                      dummy_output + kFrameLengthSamples),
+                  SetArgPointee<4>(AudioDecoder::kComfortNoise),
+                  Return(rtc::checked_cast<int>(kFrameLengthSamples))));
+  }
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+
+  // Insert 2 packets. This will make netEq into codec internal CNG mode.
+  for (int i = 0; i < 2; ++i) {
+    rtp_header.sequenceNumber += 1;
+    rtp_header.timestamp += kFrameLengthSamples;
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  }
+
+  // Pull audio.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kCNG, output.speech_type_);
+
+  // Pull audio again. Decoder fails.
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  // We are not expecting anything for output.speech_type_, since an error was
+  // returned.
+
+  // Pull audio again, should resume codec CNG.
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kCNG, output.speech_type_);
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
+// Tests that the return value from last_output_sample_rate_hz() is equal to the
+// configured inital sample rate.
+TEST_F(NetEqImplTest, InitialLastOutputSampleRate) {
+  UseNoMocks();
+  config_.sample_rate_hz = 48000;
+  CreateInstance();
+  EXPECT_EQ(48000, neteq_->last_output_sample_rate_hz());
+}
+
+TEST_F(NetEqImplTest, TickTimerIncrement) {
+  UseNoMocks();
+  CreateInstance();
+  ASSERT_TRUE(tick_timer_);
+  EXPECT_EQ(0u, tick_timer_->ticks());
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  EXPECT_EQ(1u, tick_timer_->ticks());
+}
+
+TEST_F(NetEqImplTest, SetBaseMinimumDelay) {
+  UseNoMocks();
+  use_mock_neteq_controller_ = true;
+  CreateInstance();
+
+  EXPECT_CALL(*mock_neteq_controller_, SetBaseMinimumDelay(_))
+      .WillOnce(Return(true))
+      .WillOnce(Return(false));
+
+  const int delay_ms = 200;
+
+  EXPECT_EQ(true, neteq_->SetBaseMinimumDelayMs(delay_ms));
+  EXPECT_EQ(false, neteq_->SetBaseMinimumDelayMs(delay_ms));
+}
+
+TEST_F(NetEqImplTest, GetBaseMinimumDelayMs) {
+  UseNoMocks();
+  use_mock_neteq_controller_ = true;
+  CreateInstance();
+
+  const int delay_ms = 200;
+
+  EXPECT_CALL(*mock_neteq_controller_, GetBaseMinimumDelay())
+      .WillOnce(Return(delay_ms));
+
+  EXPECT_EQ(delay_ms, neteq_->GetBaseMinimumDelayMs());
+}
+
+TEST_F(NetEqImplTest, TargetDelayMs) {
+  UseNoMocks();
+  use_mock_neteq_controller_ = true;
+  CreateInstance();
+  constexpr int kTargetLevelMs = 510;
+  EXPECT_CALL(*mock_neteq_controller_, TargetLevelMs())
+      .WillOnce(Return(kTargetLevelMs));
+  EXPECT_EQ(510, neteq_->TargetDelayMs());
+}
+
+TEST_F(NetEqImplTest, InsertEmptyPacket) {
+  UseNoMocks();
+  use_mock_neteq_controller_ = true;
+  CreateInstance();
+
+  RTPHeader rtp_header;
+  rtp_header.payloadType = 17;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_CALL(*mock_neteq_controller_, RegisterEmptyPacket());
+  neteq_->InsertEmptyPacket(rtp_header);
+}
+
+TEST_F(NetEqImplTest, NotifyControllerOfReorderedPacket) {
+  using ::testing::AllOf;
+  using ::testing::Field;
+  UseNoMocks();
+  use_mock_neteq_controller_ = true;
+  CreateInstance();
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .Times(1)
+      .WillOnce(Return(NetEq::Operation::kNormal));
+
+  const int kPayloadLengthSamples = 80;
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;  // PCM 16-bit.
+  const uint8_t kPayloadType = 17;  // Just an arbitrary number.
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+
+  // Insert second packet that was sent before the first packet.
+  rtp_header.sequenceNumber -= 1;
+  rtp_header.timestamp -= kPayloadLengthSamples;
+  EXPECT_CALL(
+      *mock_neteq_controller_,
+      PacketArrived(
+          /*fs_hz*/ 8000,
+          /*should_update_stats*/ true,
+          /*info*/
+          AllOf(
+              Field(&NetEqController::PacketArrivedInfo::packet_length_samples,
+                    kPayloadLengthSamples),
+              Field(&NetEqController::PacketArrivedInfo::main_sequence_number,
+                    rtp_header.sequenceNumber),
+              Field(&NetEqController::PacketArrivedInfo::main_timestamp,
+                    rtp_header.timestamp))));
+
+  EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+}
+
+// When using a codec with 1000 channels, there should be no crashes.
+TEST_F(NetEqImplTest, NoCrashWith1000Channels) {
+  using ::testing::AllOf;
+  using ::testing::Field;
+  UseNoMocks();
+  use_mock_decoder_database_ = true;
+  enable_muted_state_ = true;
+  CreateInstance();
+  const size_t kPayloadLength = 100;
+  const uint8_t kPayloadType = 0;
+  const uint16_t kFirstSequenceNumber = 0x1234;
+  const uint32_t kFirstTimestamp = 0x12345678;
+  const uint32_t kSsrc = 0x87654321;
+  uint8_t payload[kPayloadLength] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = kFirstSequenceNumber;
+  rtp_header.timestamp = kFirstTimestamp;
+  rtp_header.ssrc = kSsrc;
+  Packet fake_packet;
+  fake_packet.payload_type = kPayloadType;
+  fake_packet.sequence_number = kFirstSequenceNumber;
+  fake_packet.timestamp = kFirstTimestamp;
+
+  AudioDecoder* decoder = nullptr;
+
+  auto mock_decoder_factory = rtc::make_ref_counted<MockAudioDecoderFactory>();
+  EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _, _))
+      .WillOnce(Invoke([&](const SdpAudioFormat& format,
+                           absl::optional<AudioCodecPairId> codec_pair_id,
+                           std::unique_ptr<AudioDecoder>* dec) {
+        EXPECT_EQ("pcmu", format.name);
+        *dec = std::make_unique<AudioDecoderPcmU>(1000);
+        decoder = dec->get();
+      }));
+  DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                    absl::nullopt, mock_decoder_factory.get());
+  // Expectations for decoder database.
+  EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
+      .WillRepeatedly(Return(&info));
+  EXPECT_CALL(*mock_decoder_database_, GetActiveCngDecoder())
+      .WillRepeatedly(ReturnNull());
+  EXPECT_CALL(*mock_decoder_database_, GetActiveDecoder())
+      .WillRepeatedly(Return(decoder));
+  EXPECT_CALL(*mock_decoder_database_, SetActiveDecoder(_, _))
+      .WillOnce(Invoke([](uint8_t rtp_payload_type, bool* new_decoder) {
+        *new_decoder = true;
+        return 0;
+      }));
+
+  // Insert first packet.
+  neteq_->InsertPacket(rtp_header, payload);
+
+  AudioFrame audio_frame;
+  bool muted;
+
+  // Repeat 40 times to ensure we enter muted state.
+  for (int i = 0; i < 40; i++) {
+    // GetAudio should return an error, and not crash, even in muted state.
+    EXPECT_NE(0, neteq_->GetAudio(&audio_frame, &muted));
+  }
+}
+
+class Decoder120ms : public AudioDecoder {
+ public:
+  Decoder120ms(int sample_rate_hz, SpeechType speech_type)
+      : sample_rate_hz_(sample_rate_hz),
+        next_value_(1),
+        speech_type_(speech_type) {}
+
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override {
+    EXPECT_EQ(sample_rate_hz_, sample_rate_hz);
+    size_t decoded_len =
+        rtc::CheckedDivExact(sample_rate_hz, 1000) * 120 * Channels();
+    for (size_t i = 0; i < decoded_len; ++i) {
+      decoded[i] = next_value_++;
+    }
+    *speech_type = speech_type_;
+    return rtc::checked_cast<int>(decoded_len);
+  }
+
+  void Reset() override { next_value_ = 1; }
+  int SampleRateHz() const override { return sample_rate_hz_; }
+  size_t Channels() const override { return 2; }
+
+ private:
+  int sample_rate_hz_;
+  int16_t next_value_;
+  SpeechType speech_type_;
+};
+
+class NetEqImplTest120ms : public NetEqImplTest {
+ protected:
+  NetEqImplTest120ms() : NetEqImplTest() {}
+  virtual ~NetEqImplTest120ms() {}
+
+  void CreateInstanceNoMocks() {
+    UseNoMocks();
+    CreateInstance(decoder_factory_);
+    EXPECT_TRUE(neteq_->RegisterPayloadType(
+        kPayloadType, SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}})));
+  }
+
+  void CreateInstanceWithDelayManagerMock() {
+    UseNoMocks();
+    use_mock_neteq_controller_ = true;
+    CreateInstance(decoder_factory_);
+    EXPECT_TRUE(neteq_->RegisterPayloadType(
+        kPayloadType, SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}})));
+  }
+
+  uint32_t timestamp_diff_between_packets() const {
+    return rtc::CheckedDivExact(kSamplingFreq_, 1000u) * 120;
+  }
+
+  uint32_t first_timestamp() const { return 10u; }
+
+  void GetFirstPacket() {
+    bool muted;
+    for (int i = 0; i < 12; i++) {
+      EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+      EXPECT_FALSE(muted);
+    }
+  }
+
+  void InsertPacket(uint32_t timestamp) {
+    RTPHeader rtp_header;
+    rtp_header.payloadType = kPayloadType;
+    rtp_header.sequenceNumber = sequence_number_;
+    rtp_header.timestamp = timestamp;
+    rtp_header.ssrc = 15;
+    const size_t kPayloadLengthBytes = 1;  // This can be arbitrary.
+    uint8_t payload[kPayloadLengthBytes] = {0};
+    EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload));
+    sequence_number_++;
+  }
+
+  void Register120msCodec(AudioDecoder::SpeechType speech_type) {
+    const uint32_t sampling_freq = kSamplingFreq_;
+    decoder_factory_ = rtc::make_ref_counted<test::FunctionAudioDecoderFactory>(
+        [sampling_freq, speech_type]() {
+          std::unique_ptr<AudioDecoder> decoder =
+              std::make_unique<Decoder120ms>(sampling_freq, speech_type);
+          RTC_CHECK_EQ(2, decoder->Channels());
+          return decoder;
+        });
+  }
+
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+  AudioFrame output_;
+  const uint32_t kPayloadType = 17;
+  const uint32_t kSamplingFreq_ = 48000;
+  uint16_t sequence_number_ = 1;
+};
+
+TEST_F(NetEqImplTest120ms, CodecInternalCng) {
+  Register120msCodec(AudioDecoder::kComfortNoise);
+  CreateInstanceNoMocks();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kCodecInternalCng,
+            neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Normal) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceNoMocks();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  EXPECT_EQ(NetEq::Operation::kNormal, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Merge) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceWithDelayManagerMock();
+
+  EXPECT_CALL(*mock_neteq_controller_, CngOff()).WillRepeatedly(Return(true));
+  InsertPacket(first_timestamp());
+
+  GetFirstPacket();
+  bool muted;
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .WillOnce(Return(NetEq::Operation::kExpand));
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+
+  InsertPacket(first_timestamp() + 2 * timestamp_diff_between_packets());
+
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .WillOnce(Return(NetEq::Operation::kMerge));
+
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kMerge, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Expand) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceNoMocks();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kExpand, neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, FastAccelerate) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceWithDelayManagerMock();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+  InsertPacket(first_timestamp() + timestamp_diff_between_packets());
+
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .Times(1)
+      .WillOnce(Return(NetEq::Operation::kFastAccelerate));
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kFastAccelerate,
+            neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, PreemptiveExpand) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceWithDelayManagerMock();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  InsertPacket(first_timestamp() + timestamp_diff_between_packets());
+
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .Times(1)
+      .WillOnce(Return(NetEq::Operation::kPreemptiveExpand));
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kPreemptiveExpand,
+            neteq_->last_operation_for_test());
+}
+
+TEST_F(NetEqImplTest120ms, Accelerate) {
+  Register120msCodec(AudioDecoder::kSpeech);
+  CreateInstanceWithDelayManagerMock();
+
+  InsertPacket(first_timestamp());
+  GetFirstPacket();
+
+  InsertPacket(first_timestamp() + timestamp_diff_between_packets());
+
+  EXPECT_CALL(*mock_neteq_controller_, GetDecision(_, _))
+      .Times(1)
+      .WillOnce(Return(NetEq::Operation::kAccelerate));
+
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
+  EXPECT_EQ(NetEq::Operation::kAccelerate, neteq_->last_operation_for_test());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
new file mode 100644
index 0000000000..a669ad727e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -0,0 +1,345 @@
+/*
+ *  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 <memory>
+
+#include "absl/memory/memory.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "system_wrappers/include/clock.h"
+#include "test/audio_decoder_proxy_factory.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+std::unique_ptr<NetEq> CreateNetEq(
+    const NetEq::Config& config,
+    Clock* clock,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+  return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock);
+}
+
+}  // namespace
+
+using ::testing::_;
+using ::testing::Return;
+using ::testing::SetArgPointee;
+
+class MockAudioDecoder final : public AudioDecoder {
+ public:
+  static const int kPacketDuration = 960;  // 48 kHz * 20 ms
+
+  MockAudioDecoder(int sample_rate_hz, size_t num_channels)
+      : sample_rate_hz_(sample_rate_hz),
+        num_channels_(num_channels),
+        fec_enabled_(false) {}
+  ~MockAudioDecoder() override { Die(); }
+  MOCK_METHOD(void, Die, ());
+
+  MOCK_METHOD(void, Reset, (), (override));
+
+  class MockFrame : public AudioDecoder::EncodedAudioFrame {
+   public:
+    MockFrame(size_t num_channels) : num_channels_(num_channels) {}
+
+    size_t Duration() const override { return kPacketDuration; }
+
+    absl::optional<DecodeResult> Decode(
+        rtc::ArrayView<int16_t> decoded) const override {
+      const size_t output_size =
+          sizeof(int16_t) * kPacketDuration * num_channels_;
+      if (decoded.size() >= output_size) {
+        memset(decoded.data(), 0,
+               sizeof(int16_t) * kPacketDuration * num_channels_);
+        return DecodeResult{kPacketDuration * num_channels_, kSpeech};
+      } else {
+        ADD_FAILURE() << "Expected decoded.size() to be >= output_size ("
+                      << decoded.size() << " vs. " << output_size << ")";
+        return absl::nullopt;
+      }
+    }
+
+   private:
+    const size_t num_channels_;
+  };
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override {
+    std::vector<ParseResult> results;
+    if (fec_enabled_) {
+      std::unique_ptr<MockFrame> fec_frame(new MockFrame(num_channels_));
+      results.emplace_back(timestamp - kPacketDuration, 1,
+                           std::move(fec_frame));
+    }
+
+    std::unique_ptr<MockFrame> frame(new MockFrame(num_channels_));
+    results.emplace_back(timestamp, 0, std::move(frame));
+    return results;
+  }
+
+  int PacketDuration(const uint8_t* encoded,
+                     size_t encoded_len) const override {
+    ADD_FAILURE() << "Since going through ParsePayload, PacketDuration should "
+                     "never get called.";
+    return kPacketDuration;
+  }
+
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override {
+    ADD_FAILURE() << "Since going through ParsePayload, PacketHasFec should "
+                     "never get called.";
+    return fec_enabled_;
+  }
+
+  int SampleRateHz() const override { return sample_rate_hz_; }
+
+  size_t Channels() const override { return num_channels_; }
+
+  void set_fec_enabled(bool enable_fec) { fec_enabled_ = enable_fec; }
+
+  bool fec_enabled() const { return fec_enabled_; }
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override {
+    ADD_FAILURE() << "Since going through ParsePayload, DecodeInternal should "
+                     "never get called.";
+    return -1;
+  }
+
+ private:
+  const int sample_rate_hz_;
+  const size_t num_channels_;
+  bool fec_enabled_;
+};
+
+class NetEqNetworkStatsTest {
+ public:
+  static const int kPayloadSizeByte = 30;
+  static const int kFrameSizeMs = 20;
+  static const uint8_t kPayloadType = 95;
+  static const int kOutputLengthMs = 10;
+
+  enum logic {
+    kIgnore,
+    kEqual,
+    kSmallerThan,
+    kLargerThan,
+  };
+
+  struct NetEqNetworkStatsCheck {
+    logic current_buffer_size_ms;
+    logic preferred_buffer_size_ms;
+    logic jitter_peaks_found;
+    logic packet_loss_rate;
+    logic expand_rate;
+    logic speech_expand_rate;
+    logic preemptive_rate;
+    logic accelerate_rate;
+    logic secondary_decoded_rate;
+    logic secondary_discarded_rate;
+    logic added_zero_samples;
+    NetEqNetworkStatistics stats_ref;
+  };
+
+  NetEqNetworkStatsTest(const SdpAudioFormat& format, MockAudioDecoder* decoder)
+      : decoder_(decoder),
+        decoder_factory_(
+            rtc::make_ref_counted<AudioDecoderProxyFactory>(decoder)),
+        samples_per_ms_(format.clockrate_hz / 1000),
+        frame_size_samples_(kFrameSizeMs * samples_per_ms_),
+        rtp_generator_(new RtpGenerator(samples_per_ms_)),
+        last_lost_time_(0),
+        packet_loss_interval_(0xffffffff) {
+    NetEq::Config config;
+    config.sample_rate_hz = format.clockrate_hz;
+    neteq_ = CreateNetEq(config, Clock::GetRealTimeClock(), decoder_factory_);
+    neteq_->RegisterPayloadType(kPayloadType, format);
+  }
+
+  bool Lost(uint32_t send_time) {
+    if (send_time - last_lost_time_ >= packet_loss_interval_) {
+      last_lost_time_ = send_time;
+      return true;
+    }
+    return false;
+  }
+
+  void SetPacketLossRate(double loss_rate) {
+    packet_loss_interval_ =
+        (loss_rate >= 1e-3 ? static_cast<double>(kFrameSizeMs) / loss_rate
+                           : 0xffffffff);
+  }
+
+  // `stats_ref`
+  // expects.x = -1, do not care
+  // expects.x = 0, 'x' in current stats should equal 'x' in `stats_ref`
+  // expects.x = 1, 'x' in current stats should < 'x' in `stats_ref`
+  // expects.x = 2, 'x' in current stats should > 'x' in `stats_ref`
+  void CheckNetworkStatistics(NetEqNetworkStatsCheck expects) {
+    NetEqNetworkStatistics stats;
+    neteq_->NetworkStatistics(&stats);
+
+#define CHECK_NETEQ_NETWORK_STATS(x)           \
+  switch (expects.x) {                         \
+    case kEqual:                               \
+      EXPECT_EQ(stats.x, expects.stats_ref.x); \
+      break;                                   \
+    case kSmallerThan:                         \
+      EXPECT_LT(stats.x, expects.stats_ref.x); \
+      break;                                   \
+    case kLargerThan:                          \
+      EXPECT_GT(stats.x, expects.stats_ref.x); \
+      break;                                   \
+    default:                                   \
+      break;                                   \
+  }
+
+    CHECK_NETEQ_NETWORK_STATS(current_buffer_size_ms);
+    CHECK_NETEQ_NETWORK_STATS(preferred_buffer_size_ms);
+    CHECK_NETEQ_NETWORK_STATS(jitter_peaks_found);
+    CHECK_NETEQ_NETWORK_STATS(expand_rate);
+    CHECK_NETEQ_NETWORK_STATS(speech_expand_rate);
+    CHECK_NETEQ_NETWORK_STATS(preemptive_rate);
+    CHECK_NETEQ_NETWORK_STATS(accelerate_rate);
+    CHECK_NETEQ_NETWORK_STATS(secondary_decoded_rate);
+    CHECK_NETEQ_NETWORK_STATS(secondary_discarded_rate);
+
+#undef CHECK_NETEQ_NETWORK_STATS
+  }
+
+  void RunTest(int num_loops, NetEqNetworkStatsCheck expects) {
+    uint32_t time_now;
+    uint32_t next_send_time;
+
+    // Initiate `last_lost_time_`.
+    time_now = next_send_time = last_lost_time_ = rtp_generator_->GetRtpHeader(
+        kPayloadType, frame_size_samples_, &rtp_header_);
+    for (int k = 0; k < num_loops; ++k) {
+      // Delay by one frame such that the FEC can come in.
+      while (time_now + kFrameSizeMs >= next_send_time) {
+        next_send_time = rtp_generator_->GetRtpHeader(
+            kPayloadType, frame_size_samples_, &rtp_header_);
+        if (!Lost(next_send_time)) {
+          static const uint8_t payload[kPayloadSizeByte] = {0};
+          ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header_, payload));
+        }
+      }
+      bool muted = true;
+      EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_frame_, &muted));
+      ASSERT_FALSE(muted);
+      EXPECT_EQ(decoder_->Channels(), output_frame_.num_channels_);
+      EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * samples_per_ms_),
+                output_frame_.samples_per_channel_);
+      EXPECT_EQ(48000, neteq_->last_output_sample_rate_hz());
+
+      time_now += kOutputLengthMs;
+    }
+    CheckNetworkStatistics(expects);
+    neteq_->FlushBuffers();
+  }
+
+  void DecodeFecTest() {
+    decoder_->set_fec_enabled(false);
+    NetEqNetworkStatsCheck expects = {kIgnore,  // current_buffer_size_ms
+                                      kIgnore,  // preferred_buffer_size_ms
+                                      kIgnore,  // jitter_peaks_found
+                                      kEqual,   // packet_loss_rate
+                                      kEqual,   // expand_rate
+                                      kEqual,   // voice_expand_rate
+                                      kIgnore,  // preemptive_rate
+                                      kEqual,   // accelerate_rate
+                                      kEqual,   // decoded_fec_rate
+                                      kEqual,   // discarded_fec_rate
+                                      kEqual,   // added_zero_samples
+                                      {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}};
+    RunTest(50, expects);
+
+    // Next we introduce packet losses.
+    SetPacketLossRate(0.1);
+    expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 898;
+    RunTest(50, expects);
+
+    // Next we enable FEC.
+    decoder_->set_fec_enabled(true);
+    // If FEC fills in the lost packets, no packet loss will be counted.
+    expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 0;
+    expects.stats_ref.secondary_decoded_rate = 2006;
+    expects.stats_ref.secondary_discarded_rate = 14336;
+    RunTest(50, expects);
+  }
+
+  void NoiseExpansionTest() {
+    NetEqNetworkStatsCheck expects = {kIgnore,  // current_buffer_size_ms
+                                      kIgnore,  // preferred_buffer_size_ms
+                                      kIgnore,  // jitter_peaks_found
+                                      kEqual,   // packet_loss_rate
+                                      kEqual,   // expand_rate
+                                      kEqual,   // speech_expand_rate
+                                      kIgnore,  // preemptive_rate
+                                      kEqual,   // accelerate_rate
+                                      kEqual,   // decoded_fec_rate
+                                      kEqual,   // discard_fec_rate
+                                      kEqual,   // added_zero_samples
+                                      {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}};
+    RunTest(50, expects);
+
+    SetPacketLossRate(1);
+    expects.stats_ref.expand_rate = 16384;
+    expects.stats_ref.speech_expand_rate = 5324;
+    RunTest(10, expects);  // Lost 10 * 20ms in a row.
+  }
+
+ private:
+  MockAudioDecoder* decoder_;
+  rtc::scoped_refptr<AudioDecoderProxyFactory> decoder_factory_;
+  std::unique_ptr<NetEq> neteq_;
+
+  const int samples_per_ms_;
+  const size_t frame_size_samples_;
+  std::unique_ptr<RtpGenerator> rtp_generator_;
+  RTPHeader rtp_header_;
+  uint32_t last_lost_time_;
+  uint32_t packet_loss_interval_;
+  AudioFrame output_frame_;
+};
+
+TEST(NetEqNetworkStatsTest, DecodeFec) {
+  MockAudioDecoder decoder(48000, 1);
+  NetEqNetworkStatsTest test(SdpAudioFormat("opus", 48000, 2), &decoder);
+  test.DecodeFecTest();
+  EXPECT_CALL(decoder, Die()).Times(1);
+}
+
+TEST(NetEqNetworkStatsTest, StereoDecodeFec) {
+  MockAudioDecoder decoder(48000, 2);
+  NetEqNetworkStatsTest test(SdpAudioFormat("opus", 48000, 2), &decoder);
+  test.DecodeFecTest();
+  EXPECT_CALL(decoder, Die()).Times(1);
+}
+
+TEST(NetEqNetworkStatsTest, NoiseExpansionTest) {
+  MockAudioDecoder decoder(48000, 1);
+  NetEqNetworkStatsTest test(SdpAudioFormat("opus", 48000, 2), &decoder);
+  test.NoiseExpansionTest();
+  EXPECT_CALL(decoder, Die()).Times(1);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
new file mode 100644
index 0000000000..6fa56fd1c1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -0,0 +1,424 @@
+/*
+ *  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.
+ */
+
+// Test to verify correct stereo and multi-channel operation.
+
+#include <algorithm>
+#include <list>
+#include <memory>
+#include <string>
+
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+struct TestParameters {
+  int frame_size;
+  int sample_rate;
+  size_t num_channels;
+};
+
+// This is a parameterized test. The test parameters are supplied through a
+// TestParameters struct, which is obtained through the GetParam() method.
+//
+// The objective of the test is to create a mono input signal and a
+// multi-channel input signal, where each channel is identical to the mono
+// input channel. The two input signals are processed through their respective
+// NetEq instances. After that, the output signals are compared. The expected
+// result is that each channel in the multi-channel output is identical to the
+// mono output.
+class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
+ protected:
+  static const int kTimeStepMs = 10;
+  static const size_t kMaxBlockSize = 480;  // 10 ms @ 48 kHz.
+  static const uint8_t kPayloadTypeMono = 95;
+  static const uint8_t kPayloadTypeMulti = 96;
+
+  NetEqStereoTest()
+      : num_channels_(GetParam().num_channels),
+        sample_rate_hz_(GetParam().sample_rate),
+        samples_per_ms_(sample_rate_hz_ / 1000),
+        frame_size_ms_(GetParam().frame_size),
+        frame_size_samples_(
+            static_cast<size_t>(frame_size_ms_ * samples_per_ms_)),
+        output_size_samples_(10 * samples_per_ms_),
+        clock_(0),
+        rtp_generator_mono_(samples_per_ms_),
+        rtp_generator_(samples_per_ms_),
+        payload_size_bytes_(0),
+        multi_payload_size_bytes_(0),
+        last_send_time_(0),
+        last_arrival_time_(0) {
+    NetEq::Config config;
+    config.sample_rate_hz = sample_rate_hz_;
+    DefaultNetEqFactory neteq_factory;
+    auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+    neteq_mono_ = neteq_factory.CreateNetEq(config, decoder_factory, &clock_);
+    neteq_ = neteq_factory.CreateNetEq(config, decoder_factory, &clock_);
+    input_ = new int16_t[frame_size_samples_];
+    encoded_ = new uint8_t[2 * frame_size_samples_];
+    input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];
+    encoded_multi_channel_ =
+        new uint8_t[frame_size_samples_ * 2 * num_channels_];
+  }
+
+  ~NetEqStereoTest() {
+    delete[] input_;
+    delete[] encoded_;
+    delete[] input_multi_channel_;
+    delete[] encoded_multi_channel_;
+  }
+
+  virtual void SetUp() {
+    const std::string file_name =
+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+    input_file_.reset(new test::InputAudioFile(file_name));
+    RTC_CHECK_GE(num_channels_, 2);
+    ASSERT_TRUE(neteq_mono_->RegisterPayloadType(
+        kPayloadTypeMono, SdpAudioFormat("l16", sample_rate_hz_, 1)));
+    ASSERT_TRUE(neteq_->RegisterPayloadType(
+        kPayloadTypeMulti,
+        SdpAudioFormat("l16", sample_rate_hz_, num_channels_)));
+  }
+
+  virtual void TearDown() {}
+
+  int GetNewPackets() {
+    if (!input_file_->Read(frame_size_samples_, input_)) {
+      return -1;
+    }
+    payload_size_bytes_ =
+        WebRtcPcm16b_Encode(input_, frame_size_samples_, encoded_);
+    if (frame_size_samples_ * 2 != payload_size_bytes_) {
+      return -1;
+    }
+    int next_send_time = rtp_generator_mono_.GetRtpHeader(
+        kPayloadTypeMono, frame_size_samples_, &rtp_header_mono_);
+    MakeMultiChannelInput();
+    multi_payload_size_bytes_ = WebRtcPcm16b_Encode(
+        input_multi_channel_, frame_size_samples_ * num_channels_,
+        encoded_multi_channel_);
+    if (frame_size_samples_ * 2 * num_channels_ != multi_payload_size_bytes_) {
+      return -1;
+    }
+    rtp_generator_.GetRtpHeader(kPayloadTypeMulti, frame_size_samples_,
+                                &rtp_header_);
+    return next_send_time;
+  }
+
+  virtual void MakeMultiChannelInput() {
+    test::InputAudioFile::DuplicateInterleaved(
+        input_, frame_size_samples_, num_channels_, input_multi_channel_);
+  }
+
+  virtual void VerifyOutput(size_t num_samples) {
+    const int16_t* output_data = output_.data();
+    const int16_t* output_multi_channel_data = output_multi_channel_.data();
+    for (size_t i = 0; i < num_samples; ++i) {
+      for (size_t j = 0; j < num_channels_; ++j) {
+        ASSERT_EQ(output_data[i],
+                  output_multi_channel_data[i * num_channels_ + j])
+            << "Diff in sample " << i << ", channel " << j << ".";
+      }
+    }
+  }
+
+  virtual int GetArrivalTime(int send_time) {
+    int arrival_time = last_arrival_time_ + (send_time - last_send_time_);
+    last_send_time_ = send_time;
+    last_arrival_time_ = arrival_time;
+    return arrival_time;
+  }
+
+  virtual bool Lost() { return false; }
+
+  void RunTest(int num_loops) {
+    // Get next input packets (mono and multi-channel).
+    int next_send_time;
+    int next_arrival_time;
+    do {
+      next_send_time = GetNewPackets();
+      ASSERT_NE(-1, next_send_time);
+      next_arrival_time = GetArrivalTime(next_send_time);
+    } while (Lost());  // If lost, immediately read the next packet.
+
+    int time_now = 0;
+    for (int k = 0; k < num_loops; ++k) {
+      while (time_now >= next_arrival_time) {
+        // Insert packet in mono instance.
+        ASSERT_EQ(NetEq::kOK,
+                  neteq_mono_->InsertPacket(
+                      rtp_header_mono_, rtc::ArrayView<const uint8_t>(
+                                            encoded_, payload_size_bytes_)));
+        // Insert packet in multi-channel instance.
+        ASSERT_EQ(NetEq::kOK, neteq_->InsertPacket(
+                                  rtp_header_, rtc::ArrayView<const uint8_t>(
+                                                   encoded_multi_channel_,
+                                                   multi_payload_size_bytes_)));
+        // Get next input packets (mono and multi-channel).
+        do {
+          next_send_time = GetNewPackets();
+          ASSERT_NE(-1, next_send_time);
+          next_arrival_time = GetArrivalTime(next_send_time);
+        } while (Lost());  // If lost, immediately read the next packet.
+      }
+      // Get audio from mono instance.
+      bool muted;
+      EXPECT_EQ(NetEq::kOK, neteq_mono_->GetAudio(&output_, &muted));
+      ASSERT_FALSE(muted);
+      EXPECT_EQ(1u, output_.num_channels_);
+      EXPECT_EQ(output_size_samples_, output_.samples_per_channel_);
+      // Get audio from multi-channel instance.
+      ASSERT_EQ(NetEq::kOK, neteq_->GetAudio(&output_multi_channel_, &muted));
+      ASSERT_FALSE(muted);
+      EXPECT_EQ(num_channels_, output_multi_channel_.num_channels_);
+      EXPECT_EQ(output_size_samples_,
+                output_multi_channel_.samples_per_channel_);
+      rtc::StringBuilder ss;
+      ss << "Lap number " << k << ".";
+      SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+      // Compare mono and multi-channel.
+      ASSERT_NO_FATAL_FAILURE(VerifyOutput(output_size_samples_));
+
+      time_now += kTimeStepMs;
+      clock_.AdvanceTimeMilliseconds(kTimeStepMs);
+    }
+  }
+
+  const size_t num_channels_;
+  const int sample_rate_hz_;
+  const int samples_per_ms_;
+  const int frame_size_ms_;
+  const size_t frame_size_samples_;
+  const size_t output_size_samples_;
+  SimulatedClock clock_;
+  std::unique_ptr<NetEq> neteq_mono_;
+  std::unique_ptr<NetEq> neteq_;
+  test::RtpGenerator rtp_generator_mono_;
+  test::RtpGenerator rtp_generator_;
+  int16_t* input_;
+  int16_t* input_multi_channel_;
+  uint8_t* encoded_;
+  uint8_t* encoded_multi_channel_;
+  AudioFrame output_;
+  AudioFrame output_multi_channel_;
+  RTPHeader rtp_header_mono_;
+  RTPHeader rtp_header_;
+  size_t payload_size_bytes_;
+  size_t multi_payload_size_bytes_;
+  int last_send_time_;
+  int last_arrival_time_;
+  std::unique_ptr<test::InputAudioFile> input_file_;
+};
+
+class NetEqStereoTestNoJitter : public NetEqStereoTest {
+ protected:
+  NetEqStereoTestNoJitter() : NetEqStereoTest() {
+    // Start the sender 100 ms before the receiver to pre-fill the buffer.
+    // This is to avoid doing preemptive expand early in the test.
+    // TODO(hlundin): Mock the decision making instead to control the modes.
+    last_arrival_time_ = -100;
+  }
+};
+
+TEST_P(NetEqStereoTestNoJitter, RunTest) {
+  RunTest(8);
+}
+
+class NetEqStereoTestPositiveDrift : public NetEqStereoTest {
+ protected:
+  NetEqStereoTestPositiveDrift() : NetEqStereoTest(), drift_factor(0.9) {
+    // Start the sender 100 ms before the receiver to pre-fill the buffer.
+    // This is to avoid doing preemptive expand early in the test.
+    // TODO(hlundin): Mock the decision making instead to control the modes.
+    last_arrival_time_ = -100;
+  }
+  virtual int GetArrivalTime(int send_time) {
+    int arrival_time =
+        last_arrival_time_ + drift_factor * (send_time - last_send_time_);
+    last_send_time_ = send_time;
+    last_arrival_time_ = arrival_time;
+    return arrival_time;
+  }
+
+  double drift_factor;
+};
+
+TEST_P(NetEqStereoTestPositiveDrift, RunTest) {
+  RunTest(100);
+}
+
+class NetEqStereoTestNegativeDrift : public NetEqStereoTestPositiveDrift {
+ protected:
+  NetEqStereoTestNegativeDrift() : NetEqStereoTestPositiveDrift() {
+    drift_factor = 1.1;
+    last_arrival_time_ = 0;
+  }
+};
+
+TEST_P(NetEqStereoTestNegativeDrift, RunTest) {
+  RunTest(100);
+}
+
+class NetEqStereoTestDelays : public NetEqStereoTest {
+ protected:
+  static const int kDelayInterval = 10;
+  static const int kDelay = 1000;
+  NetEqStereoTestDelays() : NetEqStereoTest(), frame_index_(0) {}
+
+  virtual int GetArrivalTime(int send_time) {
+    // Deliver immediately, unless we have a back-log.
+    int arrival_time = std::min(last_arrival_time_, send_time);
+    if (++frame_index_ % kDelayInterval == 0) {
+      // Delay this packet.
+      arrival_time += kDelay;
+    }
+    last_send_time_ = send_time;
+    last_arrival_time_ = arrival_time;
+    return arrival_time;
+  }
+
+  int frame_index_;
+};
+
+TEST_P(NetEqStereoTestDelays, RunTest) {
+  RunTest(1000);
+}
+
+class NetEqStereoTestLosses : public NetEqStereoTest {
+ protected:
+  static const int kLossInterval = 10;
+  NetEqStereoTestLosses() : NetEqStereoTest(), frame_index_(0) {}
+
+  virtual bool Lost() { return (++frame_index_) % kLossInterval == 0; }
+
+  // TODO(hlundin): NetEq is not giving bitexact results for these cases.
+  virtual void VerifyOutput(size_t num_samples) {
+    for (size_t i = 0; i < num_samples; ++i) {
+      const int16_t* output_data = output_.data();
+      const int16_t* output_multi_channel_data = output_multi_channel_.data();
+      auto first_channel_sample = output_multi_channel_data[i * num_channels_];
+      for (size_t j = 0; j < num_channels_; ++j) {
+        const int kErrorMargin = 200;
+        EXPECT_NEAR(output_data[i],
+                    output_multi_channel_data[i * num_channels_ + j],
+                    kErrorMargin)
+            << "Diff in sample " << i << ", channel " << j << ".";
+        EXPECT_EQ(first_channel_sample,
+                  output_multi_channel_data[i * num_channels_ + j]);
+      }
+    }
+  }
+
+  int frame_index_;
+};
+
+TEST_P(NetEqStereoTestLosses, RunTest) {
+  RunTest(100);
+}
+
+class NetEqStereoTestSingleActiveChannelPlc : public NetEqStereoTestLosses {
+ protected:
+  NetEqStereoTestSingleActiveChannelPlc() : NetEqStereoTestLosses() {}
+
+  virtual void MakeMultiChannelInput() override {
+    // Create a multi-channel input by copying the mono channel from file to the
+    // first channel, and setting the others to zero.
+    memset(input_multi_channel_, 0,
+           frame_size_samples_ * num_channels_ * sizeof(int16_t));
+    for (size_t i = 0; i < frame_size_samples_; ++i) {
+      input_multi_channel_[i * num_channels_] = input_[i];
+    }
+  }
+
+  virtual void VerifyOutput(size_t num_samples) override {
+    // Simply verify that all samples in channels other than the first are zero.
+    const int16_t* output_multi_channel_data = output_multi_channel_.data();
+    for (size_t i = 0; i < num_samples; ++i) {
+      for (size_t j = 1; j < num_channels_; ++j) {
+        EXPECT_EQ(0, output_multi_channel_data[i * num_channels_ + j])
+            << "Sample " << i << ", channel " << j << " is non-zero.";
+      }
+    }
+  }
+};
+
+TEST_P(NetEqStereoTestSingleActiveChannelPlc, RunTest) {
+  RunTest(100);
+}
+
+// Creates a list of parameter sets.
+std::list<TestParameters> GetTestParameters() {
+  std::list<TestParameters> l;
+  const int sample_rates[] = {8000, 16000, 32000};
+  const int num_rates = sizeof(sample_rates) / sizeof(sample_rates[0]);
+  // Loop through sample rates.
+  for (int rate_index = 0; rate_index < num_rates; ++rate_index) {
+    int sample_rate = sample_rates[rate_index];
+    // Loop through all frame sizes between 10 and 60 ms.
+    for (int frame_size = 10; frame_size <= 60; frame_size += 10) {
+      TestParameters p;
+      p.frame_size = frame_size;
+      p.sample_rate = sample_rate;
+      p.num_channels = 2;
+      l.push_back(p);
+      if (sample_rate == 8000) {
+        // Add a five-channel test for 8000 Hz.
+        p.num_channels = 5;
+        l.push_back(p);
+      }
+    }
+  }
+  return l;
+}
+
+// Pretty-printing the test parameters in case of an error.
+void PrintTo(const TestParameters& p, ::std::ostream* os) {
+  *os << "{frame_size = " << p.frame_size
+      << ", num_channels = " << p.num_channels
+      << ", sample_rate = " << p.sample_rate << "}";
+}
+
+// Instantiate the tests. Each test is instantiated using the function above,
+// so that all different parameter combinations are tested.
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestNoJitter,
+                         ::testing::ValuesIn(GetTestParameters()));
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestPositiveDrift,
+                         ::testing::ValuesIn(GetTestParameters()));
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestNegativeDrift,
+                         ::testing::ValuesIn(GetTestParameters()));
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestDelays,
+                         ::testing::ValuesIn(GetTestParameters()));
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestLosses,
+                         ::testing::ValuesIn(GetTestParameters()));
+
+INSTANTIATE_TEST_SUITE_P(MultiChannel,
+                         NetEqStereoTestSingleActiveChannelPlc,
+                         ::testing::ValuesIn(GetTestParameters()));
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc
new file mode 100644
index 0000000000..fff14bf830
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -0,0 +1,1009 @@
+/*
+ *  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 "api/neteq/neteq.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>  // memset
+
+#include <algorithm>
+#include <memory>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/test/neteq_decoding_test.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/include/rtcp_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/message_digest.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/arch.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(bool, gen_ref, false, "Generate reference files.");
+
+namespace webrtc {
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) &&         \
+    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
+    (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
+    defined(WEBRTC_CODEC_ILBC)
+#define MAYBE_TestBitExactness TestBitExactness
+#else
+#define MAYBE_TestBitExactness DISABLED_TestBitExactness
+#endif
+TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
+
+  const std::string output_checksum =
+      "dee7a10ab92526876a70a85bc48a4906901af3df";
+
+  const std::string network_stats_checksum =
+      "911dbf5fd97f48d25b8f0967286eb73c9d6f6158";
+
+  DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
+                   absl::GetFlag(FLAGS_gen_ref));
+}
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && \
+    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && defined(WEBRTC_CODEC_OPUS)
+#define MAYBE_TestOpusBitExactness TestOpusBitExactness
+#else
+#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
+#endif
+TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
+
+  const std::string output_checksum =
+      "fec6827bb9ee0b21770bbbb4a3a6f8823bf537dc|"
+      "3610cc7be4b3407b9c273b1299ab7f8f47cca96b";
+
+  const std::string network_stats_checksum =
+      "3d043e47e5f4bb81d37e7bce8c44bf802965c853|"
+      "076662525572dba753b11578330bd491923f7f5e";
+
+  DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
+                   absl::GetFlag(FLAGS_gen_ref));
+}
+
+#if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && \
+    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && defined(WEBRTC_CODEC_OPUS)
+#define MAYBE_TestOpusDtxBitExactness TestOpusDtxBitExactness
+#else
+#define MAYBE_TestOpusDtxBitExactness DISABLED_TestOpusDtxBitExactness
+#endif
+TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp");
+
+  const std::string output_checksum =
+      "b3c4899eab5378ef5e54f2302948872149f6ad5e|"
+      "589e975ec31ea13f302457fea1425be9380ffb96";
+
+  const std::string network_stats_checksum =
+      "dc8447b9fee1a21fd5d1f4045d62b982a3fb0215";
+
+  DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
+                   absl::GetFlag(FLAGS_gen_ref));
+}
+
+// Use fax mode to avoid time-scaling. This is to simplify the testing of
+// packet waiting times in the packet buffer.
+class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
+ protected:
+  NetEqDecodingTestFaxMode() : NetEqDecodingTest() {
+    config_.for_test_no_time_stretching = true;
+  }
+  void TestJitterBufferDelay(bool apply_packet_loss);
+};
+
+TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
+  // Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
+  size_t num_frames = 30;
+  const size_t kSamples = 10 * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  for (size_t i = 0; i < num_frames; ++i) {
+    const uint8_t payload[kPayloadBytes] = {0};
+    RTPHeader rtp_info;
+    rtp_info.sequenceNumber = rtc::checked_cast<uint16_t>(i);
+    rtp_info.timestamp = rtc::checked_cast<uint32_t>(i * kSamples);
+    rtp_info.ssrc = 0x1234;     // Just an arbitrary SSRC.
+    rtp_info.payloadType = 94;  // PCM16b WB codec.
+    rtp_info.markerBit = 0;
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+  }
+  // Pull out all data.
+  for (size_t i = 0; i < num_frames; ++i) {
+    bool muted;
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
+  // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
+  // spacing (per definition), we expect the delay to increase with 10 ms for
+  // each packet. Thus, we are calculating the statistics for a series from 10
+  // to 300, in steps of 10 ms.
+  EXPECT_EQ(155, stats.mean_waiting_time_ms);
+  EXPECT_EQ(155, stats.median_waiting_time_ms);
+  EXPECT_EQ(10, stats.min_waiting_time_ms);
+  EXPECT_EQ(300, stats.max_waiting_time_ms);
+
+  // Check statistics again and make sure it's been reset.
+  EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
+  EXPECT_EQ(-1, stats.mean_waiting_time_ms);
+  EXPECT_EQ(-1, stats.median_waiting_time_ms);
+  EXPECT_EQ(-1, stats.min_waiting_time_ms);
+  EXPECT_EQ(-1, stats.max_waiting_time_ms);
+}
+
+
+TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDrift) {
+  // Apply a clock drift of -25 ms / s (sender faster than receiver).
+  const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
+  const double kNetworkFreezeTimeMs = 0.0;
+  const bool kGetAudioDuringFreezeRecovery = false;
+  const int kDelayToleranceMs = 20;
+  const int kMaxTimeToSpeechMs = 100;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDrift) {
+  // Apply a clock drift of +25 ms / s (sender slower than receiver).
+  const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
+  const double kNetworkFreezeTimeMs = 0.0;
+  const bool kGetAudioDuringFreezeRecovery = false;
+  const int kDelayToleranceMs = 40;
+  const int kMaxTimeToSpeechMs = 100;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDriftNetworkFreeze) {
+  // Apply a clock drift of -25 ms / s (sender faster than receiver).
+  const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
+  const double kNetworkFreezeTimeMs = 5000.0;
+  const bool kGetAudioDuringFreezeRecovery = false;
+  const int kDelayToleranceMs = 60;
+  const int kMaxTimeToSpeechMs = 200;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreeze) {
+  // Apply a clock drift of +25 ms / s (sender slower than receiver).
+  const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
+  const double kNetworkFreezeTimeMs = 5000.0;
+  const bool kGetAudioDuringFreezeRecovery = false;
+  const int kDelayToleranceMs = 40;
+  const int kMaxTimeToSpeechMs = 100;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreezeExtraPull) {
+  // Apply a clock drift of +25 ms / s (sender slower than receiver).
+  const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
+  const double kNetworkFreezeTimeMs = 5000.0;
+  const bool kGetAudioDuringFreezeRecovery = true;
+  const int kDelayToleranceMs = 40;
+  const int kMaxTimeToSpeechMs = 100;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, LongCngWithoutClockDrift) {
+  const double kDriftFactor = 1.0;  // No drift.
+  const double kNetworkFreezeTimeMs = 0.0;
+  const bool kGetAudioDuringFreezeRecovery = false;
+  const int kDelayToleranceMs = 10;
+  const int kMaxTimeToSpeechMs = 50;
+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
+                        kMaxTimeToSpeechMs);
+}
+
+TEST_F(NetEqDecodingTest, UnknownPayloadType) {
+  const size_t kPayloadBytes = 100;
+  uint8_t payload[kPayloadBytes] = {0};
+  RTPHeader rtp_info;
+  PopulateRtpInfo(0, 0, &rtp_info);
+  rtp_info.payloadType = 1;  // Not registered as a decoder.
+  EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info, payload));
+}
+
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#define MAYBE_DecoderError DecoderError
+#else
+#define MAYBE_DecoderError DISABLED_DecoderError
+#endif
+
+TEST_F(NetEqDecodingTest, MAYBE_DecoderError) {
+  const size_t kPayloadBytes = 100;
+  uint8_t payload[kPayloadBytes] = {0};
+  RTPHeader rtp_info;
+  PopulateRtpInfo(0, 0, &rtp_info);
+  rtp_info.payloadType = 103;  // iSAC, but the payload is invalid.
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+  // Set all of `out_data_` to 1, and verify that it was set to 0 by the call
+  // to GetAudio.
+  int16_t* out_frame_data = out_frame_.mutable_data();
+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
+    out_frame_data[i] = 1;
+  }
+  bool muted;
+  EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
+
+  // Verify that the first 160 samples are set to 0.
+  static const int kExpectedOutputLength = 160;  // 10 ms at 16 kHz sample rate.
+  const int16_t* const_out_frame_data = out_frame_.data();
+  for (int i = 0; i < kExpectedOutputLength; ++i) {
+    rtc::StringBuilder ss;
+    ss << "i = " << i;
+    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+    EXPECT_EQ(0, const_out_frame_data[i]);
+  }
+}
+
+TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
+  // Set all of `out_data_` to 1, and verify that it was set to 0 by the call
+  // to GetAudio.
+  int16_t* out_frame_data = out_frame_.mutable_data();
+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
+    out_frame_data[i] = 1;
+  }
+  bool muted;
+  EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
+  // Verify that the first block of samples is set to 0.
+  static const int kExpectedOutputLength =
+      kInitSampleRateHz / 100;  // 10 ms at initial sample rate.
+  const int16_t* const_out_frame_data = out_frame_.data();
+  for (int i = 0; i < kExpectedOutputLength; ++i) {
+    rtc::StringBuilder ss;
+    ss << "i = " << i;
+    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+    EXPECT_EQ(0, const_out_frame_data[i]);
+  }
+  // Verify that the sample rate did not change from the initial configuration.
+  EXPECT_EQ(config_.sample_rate_hz, neteq_->last_output_sample_rate_hz());
+}
+
+class NetEqBgnTest : public NetEqDecodingTest {
+ protected:
+  void CheckBgn(int sampling_rate_hz) {
+    size_t expected_samples_per_channel = 0;
+    uint8_t payload_type = 0xFF;  // Invalid.
+    if (sampling_rate_hz == 8000) {
+      expected_samples_per_channel = kBlockSize8kHz;
+      payload_type = 93;  // PCM 16, 8 kHz.
+    } else if (sampling_rate_hz == 16000) {
+      expected_samples_per_channel = kBlockSize16kHz;
+      payload_type = 94;  // PCM 16, 16 kHZ.
+    } else if (sampling_rate_hz == 32000) {
+      expected_samples_per_channel = kBlockSize32kHz;
+      payload_type = 95;  // PCM 16, 32 kHz.
+    } else {
+      ASSERT_TRUE(false);  // Unsupported test case.
+    }
+
+    AudioFrame output;
+    test::AudioLoop input;
+    // We are using the same 32 kHz input file for all tests, regardless of
+    // `sampling_rate_hz`. The output may sound weird, but the test is still
+    // valid.
+    ASSERT_TRUE(input.Init(
+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
+        10 * sampling_rate_hz,  // Max 10 seconds loop length.
+        expected_samples_per_channel));
+
+    // Payload of 10 ms of PCM16 32 kHz.
+    uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
+    RTPHeader rtp_info;
+    PopulateRtpInfo(0, 0, &rtp_info);
+    rtp_info.payloadType = payload_type;
+
+    bool muted;
+    for (int n = 0; n < 10; ++n) {  // Insert few packets and get audio.
+      auto block = input.GetNextBlock();
+      ASSERT_EQ(expected_samples_per_channel, block.size());
+      size_t enc_len_bytes =
+          WebRtcPcm16b_Encode(block.data(), block.size(), payload);
+      ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
+
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, enc_len_bytes)));
+      output.Reset();
+      ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+      ASSERT_EQ(1u, output.num_channels_);
+      ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
+      ASSERT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+
+      // Next packet.
+      rtp_info.timestamp +=
+          rtc::checked_cast<uint32_t>(expected_samples_per_channel);
+      rtp_info.sequenceNumber++;
+    }
+
+    output.Reset();
+
+    // Get audio without inserting packets, expecting PLC and PLC-to-CNG. Pull
+    // one frame without checking speech-type. This is the first frame pulled
+    // without inserting any packet, and might not be labeled as PLC.
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+    ASSERT_EQ(1u, output.num_channels_);
+    ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
+
+    // To be able to test the fading of background noise we need at lease to
+    // pull 611 frames.
+    const int kFadingThreshold = 611;
+
+    // Test several CNG-to-PLC packet for the expected behavior. The number 20
+    // is arbitrary, but sufficiently large to test enough number of frames.
+    const int kNumPlcToCngTestFrames = 20;
+    bool plc_to_cng = false;
+    for (int n = 0; n < kFadingThreshold + kNumPlcToCngTestFrames; ++n) {
+      output.Reset();
+      // Set to non-zero.
+      memset(output.mutable_data(), 1, AudioFrame::kMaxDataSizeBytes);
+      ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+      ASSERT_FALSE(muted);
+      ASSERT_EQ(1u, output.num_channels_);
+      ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
+      if (output.speech_type_ == AudioFrame::kPLCCNG) {
+        plc_to_cng = true;
+        double sum_squared = 0;
+        const int16_t* output_data = output.data();
+        for (size_t k = 0;
+             k < output.num_channels_ * output.samples_per_channel_; ++k)
+          sum_squared += output_data[k] * output_data[k];
+        EXPECT_EQ(0, sum_squared);
+      } else {
+        EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
+      }
+    }
+    EXPECT_TRUE(plc_to_cng);  // Just to be sure that PLC-to-CNG has occurred.
+  }
+};
+
+TEST_F(NetEqBgnTest, RunTest) {
+  CheckBgn(8000);
+  CheckBgn(16000);
+  CheckBgn(32000);
+}
+
+TEST_F(NetEqDecodingTest, SequenceNumberWrap) {
+  // Start with a sequence number that will soon wrap.
+  std::set<uint16_t> drop_seq_numbers;  // Don't drop any packets.
+  WrapTest(0xFFFF - 10, 0, drop_seq_numbers, true, false);
+}
+
+TEST_F(NetEqDecodingTest, SequenceNumberWrapAndDrop) {
+  // Start with a sequence number that will soon wrap.
+  std::set<uint16_t> drop_seq_numbers;
+  drop_seq_numbers.insert(0xFFFF);
+  drop_seq_numbers.insert(0x0);
+  WrapTest(0xFFFF - 10, 0, drop_seq_numbers, true, false);
+}
+
+TEST_F(NetEqDecodingTest, TimestampWrap) {
+  // Start with a timestamp that will soon wrap.
+  std::set<uint16_t> drop_seq_numbers;
+  WrapTest(0, 0xFFFFFFFF - 3000, drop_seq_numbers, false, true);
+}
+
+TEST_F(NetEqDecodingTest, TimestampAndSequenceNumberWrap) {
+  // Start with a timestamp and a sequence number that will wrap at the same
+  // time.
+  std::set<uint16_t> drop_seq_numbers;
+  WrapTest(0xFFFF - 10, 0xFFFFFFFF - 5000, drop_seq_numbers, true, true);
+}
+
+TEST_F(NetEqDecodingTest, DiscardDuplicateCng) {
+  uint16_t seq_no = 0;
+  uint32_t timestamp = 0;
+  const int kFrameSizeMs = 10;
+  const int kSampleRateKhz = 16;
+  const int kSamples = kFrameSizeMs * kSampleRateKhz;
+  const size_t kPayloadBytes = kSamples * 2;
+
+  const int algorithmic_delay_samples =
+      std::max(algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
+  // Insert three speech packets. Three are needed to get the frame length
+  // correct.
+  uint8_t payload[kPayloadBytes] = {0};
+  RTPHeader rtp_info;
+  bool muted;
+  for (int i = 0; i < 3; ++i) {
+    PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+    ++seq_no;
+    timestamp += kSamples;
+
+    // Pull audio once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+  // Verify speech output.
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+
+  // Insert same CNG packet twice.
+  const int kCngPeriodMs = 100;
+  const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
+  size_t payload_len;
+  PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+  // This is the first time this CNG packet is inserted.
+  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                  payload, payload_len)));
+
+  // Pull audio once and make sure CNG is played.
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+  EXPECT_FALSE(
+      neteq_->GetPlayoutTimestamp());  // Returns empty value during CNG.
+  EXPECT_EQ(timestamp - algorithmic_delay_samples,
+            out_frame_.timestamp_ + out_frame_.samples_per_channel_);
+
+  // Insert the same CNG packet again. Note that at this point it is old, since
+  // we have already decoded the first copy of it.
+  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                  payload, payload_len)));
+
+  // Pull audio until we have played `kCngPeriodMs` of CNG. Start at 10 ms since
+  // we have already pulled out CNG once.
+  for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) {
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+    EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+    EXPECT_FALSE(
+        neteq_->GetPlayoutTimestamp());  // Returns empty value during CNG.
+    EXPECT_EQ(timestamp - algorithmic_delay_samples,
+              out_frame_.timestamp_ + out_frame_.samples_per_channel_);
+  }
+
+  ++seq_no;
+  timestamp += kCngPeriodSamples;
+  uint32_t first_speech_timestamp = timestamp;
+  // Insert speech again.
+  for (int i = 0; i < 3; ++i) {
+    PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+    ++seq_no;
+    timestamp += kSamples;
+  }
+
+  // Pull audio once and verify that the output is speech again.
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+  absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+  ASSERT_TRUE(playout_timestamp);
+  EXPECT_EQ(first_speech_timestamp + kSamples - algorithmic_delay_samples,
+            *playout_timestamp);
+}
+
+TEST_F(NetEqDecodingTest, CngFirst) {
+  uint16_t seq_no = 0;
+  uint32_t timestamp = 0;
+  const int kFrameSizeMs = 10;
+  const int kSampleRateKhz = 16;
+  const int kSamples = kFrameSizeMs * kSampleRateKhz;
+  const int kPayloadBytes = kSamples * 2;
+  const int kCngPeriodMs = 100;
+  const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
+  size_t payload_len;
+
+  uint8_t payload[kPayloadBytes] = {0};
+  RTPHeader rtp_info;
+
+  PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+  ASSERT_EQ(NetEq::kOK,
+            neteq_->InsertPacket(
+                rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len)));
+  ++seq_no;
+  timestamp += kCngPeriodSamples;
+
+  // Pull audio once and make sure CNG is played.
+  bool muted;
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+
+  // Insert some speech packets.
+  const uint32_t first_speech_timestamp = timestamp;
+  int timeout_counter = 0;
+  do {
+    ASSERT_LT(timeout_counter++, 20) << "Test timed out";
+    PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+    ++seq_no;
+    timestamp += kSamples;
+
+    // Pull audio once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  } while (!IsNewerTimestamp(out_frame_.timestamp_, first_speech_timestamp));
+  // Verify speech output.
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+}
+
+class NetEqDecodingTestWithMutedState : public NetEqDecodingTest {
+ public:
+  NetEqDecodingTestWithMutedState() : NetEqDecodingTest() {
+    config_.enable_muted_state = true;
+  }
+
+ protected:
+  static constexpr size_t kSamples = 10 * 16;
+  static constexpr size_t kPayloadBytes = kSamples * 2;
+
+  void InsertPacket(uint32_t rtp_timestamp) {
+    uint8_t payload[kPayloadBytes] = {0};
+    RTPHeader rtp_info;
+    PopulateRtpInfo(0, rtp_timestamp, &rtp_info);
+    EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+  }
+
+  void InsertCngPacket(uint32_t rtp_timestamp) {
+    uint8_t payload[kPayloadBytes] = {0};
+    RTPHeader rtp_info;
+    size_t payload_len;
+    PopulateCng(0, rtp_timestamp, &rtp_info, payload, &payload_len);
+    EXPECT_EQ(NetEq::kOK,
+              neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                 payload, payload_len)));
+  }
+
+  bool GetAudioReturnMuted() {
+    bool muted;
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    return muted;
+  }
+
+  void GetAudioUntilMuted() {
+    while (!GetAudioReturnMuted()) {
+      ASSERT_LT(counter_++, 1000) << "Test timed out";
+    }
+  }
+
+  void GetAudioUntilNormal() {
+    bool muted = false;
+    while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
+      EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+      ASSERT_LT(counter_++, 1000) << "Test timed out";
+    }
+    EXPECT_FALSE(muted);
+  }
+
+  int counter_ = 0;
+};
+
+// Verifies that NetEq goes in and out of muted state as expected.
+TEST_F(NetEqDecodingTestWithMutedState, MutedState) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+  EXPECT_TRUE(out_frame_.muted());
+
+  // Verify that output audio is not written during muted mode. Other parameters
+  // should be correct, though.
+  AudioFrame new_frame;
+  int16_t* frame_data = new_frame.mutable_data();
+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
+    frame_data[i] = 17;
+  }
+  bool muted;
+  EXPECT_EQ(0, neteq_->GetAudio(&new_frame, &muted));
+  EXPECT_TRUE(muted);
+  EXPECT_TRUE(out_frame_.muted());
+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
+    EXPECT_EQ(17, frame_data[i]);
+  }
+  EXPECT_EQ(out_frame_.timestamp_ + out_frame_.samples_per_channel_,
+            new_frame.timestamp_);
+  EXPECT_EQ(out_frame_.samples_per_channel_, new_frame.samples_per_channel_);
+  EXPECT_EQ(out_frame_.sample_rate_hz_, new_frame.sample_rate_hz_);
+  EXPECT_EQ(out_frame_.num_channels_, new_frame.num_channels_);
+  EXPECT_EQ(out_frame_.speech_type_, new_frame.speech_type_);
+  EXPECT_EQ(out_frame_.vad_activity_, new_frame.vad_activity_);
+
+  // Insert new data. Timestamp is corrected for the time elapsed since the last
+  // packet. Verify that normal operation resumes.
+  InsertPacket(kSamples * counter_);
+  GetAudioUntilNormal();
+  EXPECT_FALSE(out_frame_.muted());
+
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
+  // NetEqNetworkStatistics::expand_rate tells the fraction of samples that were
+  // concealment samples, in Q14 (16384 = 100%) .The vast majority should be
+  // concealment samples in this test.
+  EXPECT_GT(stats.expand_rate, 14000);
+  // And, it should be greater than the speech_expand_rate.
+  EXPECT_GT(stats.expand_rate, stats.speech_expand_rate);
+}
+
+// Verifies that NetEq goes out of muted state when given a delayed packet.
+TEST_F(NetEqDecodingTestWithMutedState, MutedStateDelayedPacket) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+  // Insert new data. Timestamp is only corrected for the half of the time
+  // elapsed since the last packet. That is, the new packet is delayed. Verify
+  // that normal operation resumes.
+  InsertPacket(kSamples * counter_ / 2);
+  GetAudioUntilNormal();
+}
+
+// Verifies that NetEq goes out of muted state when given a future packet.
+TEST_F(NetEqDecodingTestWithMutedState, MutedStateFuturePacket) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+  // Insert new data. Timestamp is over-corrected for the time elapsed since the
+  // last packet. That is, the new packet is too early. Verify that normal
+  // operation resumes.
+  InsertPacket(kSamples * counter_ * 2);
+  GetAudioUntilNormal();
+}
+
+// Verifies that NetEq goes out of muted state when given an old packet.
+TEST_F(NetEqDecodingTestWithMutedState, MutedStateOldPacket) {
+  // Insert one speech packet.
+  InsertPacket(0);
+  // Pull out audio once and expect it not to be muted.
+  EXPECT_FALSE(GetAudioReturnMuted());
+  // Pull data until faded out.
+  GetAudioUntilMuted();
+
+  EXPECT_NE(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+  // Insert a few packets which are older than the first packet.
+  for (int i = 0; i < 5; ++i) {
+    InsertPacket(kSamples * (i - 1000));
+  }
+  EXPECT_FALSE(GetAudioReturnMuted());
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+}
+
+// Verifies that NetEq doesn't enter muted state when CNG mode is active and the
+// packet stream is suspended for a long time.
+TEST_F(NetEqDecodingTestWithMutedState, DoNotMuteExtendedCngWithoutPackets) {
+  // Insert one CNG packet.
+  InsertCngPacket(0);
+
+  // Pull 10 seconds of audio (10 ms audio generated per lap).
+  for (int i = 0; i < 1000; ++i) {
+    bool muted;
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_FALSE(muted);
+  }
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+}
+
+// Verifies that NetEq goes back to normal after a long CNG period with the
+// packet stream suspended.
+TEST_F(NetEqDecodingTestWithMutedState, RecoverAfterExtendedCngWithoutPackets) {
+  // Insert one CNG packet.
+  InsertCngPacket(0);
+
+  // Pull 10 seconds of audio (10 ms audio generated per lap).
+  for (int i = 0; i < 1000; ++i) {
+    bool muted;
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  }
+
+  // Insert new data. Timestamp is corrected for the time elapsed since the last
+  // packet. Verify that normal operation resumes.
+  InsertPacket(kSamples * counter_);
+  GetAudioUntilNormal();
+}
+
+namespace {
+::testing::AssertionResult AudioFramesEqualExceptData(const AudioFrame& a,
+                                                      const AudioFrame& b) {
+  if (a.timestamp_ != b.timestamp_)
+    return ::testing::AssertionFailure() << "timestamp_ diff (" << a.timestamp_
+                                         << " != " << b.timestamp_ << ")";
+  if (a.sample_rate_hz_ != b.sample_rate_hz_)
+    return ::testing::AssertionFailure()
+           << "sample_rate_hz_ diff (" << a.sample_rate_hz_
+           << " != " << b.sample_rate_hz_ << ")";
+  if (a.samples_per_channel_ != b.samples_per_channel_)
+    return ::testing::AssertionFailure()
+           << "samples_per_channel_ diff (" << a.samples_per_channel_
+           << " != " << b.samples_per_channel_ << ")";
+  if (a.num_channels_ != b.num_channels_)
+    return ::testing::AssertionFailure()
+           << "num_channels_ diff (" << a.num_channels_
+           << " != " << b.num_channels_ << ")";
+  if (a.speech_type_ != b.speech_type_)
+    return ::testing::AssertionFailure()
+           << "speech_type_ diff (" << a.speech_type_
+           << " != " << b.speech_type_ << ")";
+  if (a.vad_activity_ != b.vad_activity_)
+    return ::testing::AssertionFailure()
+           << "vad_activity_ diff (" << a.vad_activity_
+           << " != " << b.vad_activity_ << ")";
+  return ::testing::AssertionSuccess();
+}
+
+::testing::AssertionResult AudioFramesEqual(const AudioFrame& a,
+                                            const AudioFrame& b) {
+  ::testing::AssertionResult res = AudioFramesEqualExceptData(a, b);
+  if (!res)
+    return res;
+  if (memcmp(a.data(), b.data(),
+             a.samples_per_channel_ * a.num_channels_ * sizeof(*a.data())) !=
+      0) {
+    return ::testing::AssertionFailure() << "data_ diff";
+  }
+  return ::testing::AssertionSuccess();
+}
+
+}  // namespace
+
+TEST_F(NetEqDecodingTestTwoInstances, CompareMutedStateOnOff) {
+  ASSERT_FALSE(config_.enable_muted_state);
+  config2_.enable_muted_state = true;
+  CreateSecondInstance();
+
+  // Insert one speech packet into both NetEqs.
+  const size_t kSamples = 10 * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  uint8_t payload[kPayloadBytes] = {0};
+  RTPHeader rtp_info;
+  PopulateRtpInfo(0, 0, &rtp_info);
+  EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+  EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload));
+
+  AudioFrame out_frame1, out_frame2;
+  bool muted;
+  for (int i = 0; i < 1000; ++i) {
+    rtc::StringBuilder ss;
+    ss << "i = " << i;
+    SCOPED_TRACE(ss.str());  // Print out the loop iterator on failure.
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
+    EXPECT_FALSE(muted);
+    EXPECT_EQ(0, neteq2_->GetAudio(&out_frame2, &muted));
+    if (muted) {
+      EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
+    } else {
+      EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
+    }
+  }
+  EXPECT_TRUE(muted);
+
+  // Insert new data. Timestamp is corrected for the time elapsed since the last
+  // packet.
+  for (int i = 0; i < 5; ++i) {
+    PopulateRtpInfo(0, kSamples * 1000 + kSamples * i, &rtp_info);
+    EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+    EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload));
+  }
+
+  int counter = 0;
+  while (out_frame1.speech_type_ != AudioFrame::kNormalSpeech) {
+    ASSERT_LT(counter++, 1000) << "Test timed out";
+    rtc::StringBuilder ss;
+    ss << "counter = " << counter;
+    SCOPED_TRACE(ss.str());  // Print out the loop iterator on failure.
+    EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
+    EXPECT_FALSE(muted);
+    EXPECT_EQ(0, neteq2_->GetAudio(&out_frame2, &muted));
+    if (muted) {
+      EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
+    } else {
+      EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
+    }
+  }
+  EXPECT_FALSE(muted);
+}
+
+TEST_F(NetEqDecodingTest, TestConcealmentEvents) {
+  const int kNumConcealmentEvents = 19;
+  const size_t kSamples = 10 * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  int seq_no = 0;
+  RTPHeader rtp_info;
+  rtp_info.ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info.payloadType = 94;  // PCM16b WB codec.
+  rtp_info.markerBit = 0;
+  const uint8_t payload[kPayloadBytes] = {0};
+  bool muted;
+
+  for (int i = 0; i < kNumConcealmentEvents; i++) {
+    // Insert some packets of 10 ms size.
+    for (int j = 0; j < 10; j++) {
+      rtp_info.sequenceNumber = seq_no++;
+      rtp_info.timestamp = rtp_info.sequenceNumber * kSamples;
+      neteq_->InsertPacket(rtp_info, payload);
+      neteq_->GetAudio(&out_frame_, &muted);
+    }
+
+    // Lose a number of packets.
+    int num_lost = 1 + i;
+    for (int j = 0; j < num_lost; j++) {
+      seq_no++;
+      neteq_->GetAudio(&out_frame_, &muted);
+    }
+  }
+
+  // Check number of concealment events.
+  NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(kNumConcealmentEvents, static_cast<int>(stats.concealment_events));
+}
+
+// Test that the jitter buffer delay stat is computed correctly.
+void NetEqDecodingTestFaxMode::TestJitterBufferDelay(bool apply_packet_loss) {
+  const int kNumPackets = 10;
+  const int kDelayInNumPackets = 2;
+  const int kPacketLenMs = 10;  // All packets are of 10 ms size.
+  const size_t kSamples = kPacketLenMs * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  RTPHeader rtp_info;
+  rtp_info.ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info.payloadType = 94;  // PCM16b WB codec.
+  rtp_info.markerBit = 0;
+  const uint8_t payload[kPayloadBytes] = {0};
+  bool muted;
+  int packets_sent = 0;
+  int packets_received = 0;
+  int expected_delay = 0;
+  int expected_target_delay = 0;
+  uint64_t expected_emitted_count = 0;
+  while (packets_received < kNumPackets) {
+    // Insert packet.
+    if (packets_sent < kNumPackets) {
+      rtp_info.sequenceNumber = packets_sent++;
+      rtp_info.timestamp = rtp_info.sequenceNumber * kSamples;
+      neteq_->InsertPacket(rtp_info, payload);
+    }
+
+    // Get packet.
+    if (packets_sent > kDelayInNumPackets) {
+      neteq_->GetAudio(&out_frame_, &muted);
+      packets_received++;
+
+      // The delay reported by the jitter buffer never exceeds
+      // the number of samples previously fetched with GetAudio
+      // (hence the min()).
+      int packets_delay = std::min(packets_received, kDelayInNumPackets + 1);
+
+      // The increase of the expected delay is the product of
+      // the current delay of the jitter buffer in ms * the
+      // number of samples that are sent for play out.
+      int current_delay_ms = packets_delay * kPacketLenMs;
+      expected_delay += current_delay_ms * kSamples;
+      expected_target_delay += neteq_->TargetDelayMs() * kSamples;
+      expected_emitted_count += kSamples;
+    }
+  }
+
+  if (apply_packet_loss) {
+    // Extra call to GetAudio to cause concealment.
+    neteq_->GetAudio(&out_frame_, &muted);
+  }
+
+  // Check jitter buffer delay.
+  NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(expected_delay,
+            rtc::checked_cast<int>(stats.jitter_buffer_delay_ms));
+  EXPECT_EQ(expected_emitted_count, stats.jitter_buffer_emitted_count);
+  EXPECT_EQ(expected_target_delay,
+            rtc::checked_cast<int>(stats.jitter_buffer_target_delay_ms));
+}
+
+TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithoutLoss) {
+  TestJitterBufferDelay(false);
+}
+
+TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithLoss) {
+  TestJitterBufferDelay(true);
+}
+
+TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithAcceleration) {
+  const int kPacketLenMs = 10;  // All packets are of 10 ms size.
+  const size_t kSamples = kPacketLenMs * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  RTPHeader rtp_info;
+  rtp_info.ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info.payloadType = 94;  // PCM16b WB codec.
+  rtp_info.markerBit = 0;
+  const uint8_t payload[kPayloadBytes] = {0};
+
+  int expected_target_delay = neteq_->TargetDelayMs() * kSamples;
+  neteq_->InsertPacket(rtp_info, payload);
+
+  bool muted;
+  neteq_->GetAudio(&out_frame_, &muted);
+
+  rtp_info.sequenceNumber += 1;
+  rtp_info.timestamp += kSamples;
+  neteq_->InsertPacket(rtp_info, payload);
+  rtp_info.sequenceNumber += 1;
+  rtp_info.timestamp += kSamples;
+  neteq_->InsertPacket(rtp_info, payload);
+
+  expected_target_delay += neteq_->TargetDelayMs() * 2 * kSamples;
+  // We have two packets in the buffer and kAccelerate operation will
+  // extract 20 ms of data.
+  neteq_->GetAudio(&out_frame_, &muted, nullptr, NetEq::Operation::kAccelerate);
+
+  // Check jitter buffer delay.
+  NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(10 * kSamples * 3, stats.jitter_buffer_delay_ms);
+  EXPECT_EQ(kSamples * 3, stats.jitter_buffer_emitted_count);
+  EXPECT_EQ(expected_target_delay,
+            rtc::checked_cast<int>(stats.jitter_buffer_target_delay_ms));
+}
+
+namespace test {
+TEST(NetEqNoTimeStretchingMode, RunTest) {
+  NetEq::Config config;
+  config.for_test_no_time_stretching = true;
+  auto codecs = NetEqTest::StandardDecoderMap();
+  NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
+      {1, kRtpExtensionAudioLevel},
+      {3, kRtpExtensionAbsoluteSendTime},
+      {5, kRtpExtensionTransportSequenceNumber},
+      {7, kRtpExtensionVideoContentType},
+      {8, kRtpExtensionVideoTiming}};
+  std::unique_ptr<NetEqInput> input(new NetEqRtpDumpInput(
+      webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"),
+      rtp_ext_map, absl::nullopt /*No SSRC filter*/));
+  std::unique_ptr<TimeLimitedNetEqInput> input_time_limit(
+      new TimeLimitedNetEqInput(std::move(input), 20000));
+  std::unique_ptr<AudioSink> output(new VoidAudioSink);
+  NetEqTest::Callbacks callbacks;
+  NetEqTest test(config, CreateBuiltinAudioDecoderFactory(), codecs,
+                 /*text_log=*/nullptr, /*neteq_factory=*/nullptr,
+                 /*input=*/std::move(input_time_limit), std::move(output),
+                 callbacks);
+  test.Run();
+  const auto stats = test.SimulationStats();
+  EXPECT_EQ(0, stats.accelerate_rate);
+  EXPECT_EQ(0, stats.preemptive_rate);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.proto b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.proto
new file mode 100644
index 0000000000..b4b4253c3d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.proto
@@ -0,0 +1,31 @@
+syntax = "proto2";
+option optimize_for = LITE_RUNTIME;
+package webrtc.neteq_unittest;
+
+message NetEqNetworkStatistics {
+  // Next field number 18.
+  optional uint32 current_buffer_size_ms = 1;
+  optional uint32 preferred_buffer_size_ms = 2;
+  optional uint32 jitter_peaks_found = 3;
+  reserved 4;  // Was packet_loss_rate.
+  reserved 5;  // Was packet_discard_rate.
+  optional uint32 expand_rate = 6;
+  optional uint32 speech_expand_rate = 7;
+  optional uint32 preemptive_rate = 8;
+  optional uint32 accelerate_rate = 9;
+  optional uint32 secondary_decoded_rate = 10;
+  optional uint32 secondary_discarded_rate = 17;
+  optional int32 clockdrift_ppm = 11;
+  reserved 12;  // Was added_zero_samples.
+  optional int32 mean_waiting_time_ms = 13;
+  optional int32 median_waiting_time_ms = 14;
+  optional int32 min_waiting_time_ms = 15;
+  optional int32 max_waiting_time_ms = 16;
+}
+
+message RtcpStatistics {
+  optional uint32 fraction_lost = 1;
+  optional uint32 cumulative_lost = 2;
+  optional uint32 extended_max_sequence_number = 3;
+  optional uint32 jitter = 4;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/normal.cc b/third_party/libwebrtc/modules/audio_coding/neteq/normal.cc
new file mode 100644
index 0000000000..461ee7fa4a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/normal.cc
@@ -0,0 +1,194 @@
+/*
+ *  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/neteq/normal.h"
+
+#include <string.h>  // memset, memcpy
+
+#include <algorithm>  // min
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+int Normal::Process(const int16_t* input,
+                    size_t length,
+                    NetEq::Mode last_mode,
+                    AudioMultiVector* output) {
+  if (length == 0) {
+    // Nothing to process.
+    output->Clear();
+    return static_cast<int>(length);
+  }
+
+  RTC_DCHECK(output->Empty());
+  // Output should be empty at this point.
+  if (length % output->Channels() != 0) {
+    // The length does not match the number of channels.
+    output->Clear();
+    return 0;
+  }
+  output->PushBackInterleaved(rtc::ArrayView<const int16_t>(input, length));
+
+  const int fs_mult = fs_hz_ / 8000;
+  RTC_DCHECK_GT(fs_mult, 0);
+  // fs_shift = log2(fs_mult), rounded down.
+  // Note that `fs_shift` is not "exact" for 48 kHz.
+  // TODO(hlundin): Investigate this further.
+  const int fs_shift = 30 - WebRtcSpl_NormW32(fs_mult);
+
+  // If last call resulted in a CodedPlc we don't need to do cross-fading but we
+  // need to report the end of the interruption once we are back to normal
+  // operation.
+  if (last_mode == NetEq::Mode::kCodecPlc) {
+    statistics_->EndExpandEvent(fs_hz_);
+  }
+
+  // Check if last RecOut call resulted in an Expand. If so, we have to take
+  // care of some cross-fading and unmuting.
+  if (last_mode == NetEq::Mode::kExpand) {
+    // Generate interpolation data using Expand.
+    // First, set Expand parameters to appropriate values.
+    expand_->SetParametersForNormalAfterExpand();
+
+    // Call Expand.
+    AudioMultiVector expanded(output->Channels());
+    expand_->Process(&expanded);
+    expand_->Reset();
+
+    size_t length_per_channel = length / output->Channels();
+    std::unique_ptr<int16_t[]> signal(new int16_t[length_per_channel]);
+    for (size_t channel_ix = 0; channel_ix < output->Channels(); ++channel_ix) {
+      // Set muting factor to the same as expand muting factor.
+      int16_t mute_factor = expand_->MuteFactor(channel_ix);
+
+      (*output)[channel_ix].CopyTo(length_per_channel, 0, signal.get());
+
+      // Find largest absolute value in new data.
+      int16_t decoded_max =
+          WebRtcSpl_MaxAbsValueW16(signal.get(), length_per_channel);
+      // Adjust muting factor if needed (to BGN level).
+      size_t energy_length =
+          std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
+      int scaling = 6 + fs_shift - WebRtcSpl_NormW32(decoded_max * decoded_max);
+      scaling = std::max(scaling, 0);  // `scaling` should always be >= 0.
+      int32_t energy = WebRtcSpl_DotProductWithScale(signal.get(), signal.get(),
+                                                     energy_length, scaling);
+      int32_t scaled_energy_length =
+          static_cast<int32_t>(energy_length >> scaling);
+      if (scaled_energy_length > 0) {
+        energy = energy / scaled_energy_length;
+      } else {
+        energy = 0;
+      }
+
+      int local_mute_factor = 16384;  // 1.0 in Q14.
+      if ((energy != 0) && (energy > background_noise_.Energy(channel_ix))) {
+        // Normalize new frame energy to 15 bits.
+        scaling = WebRtcSpl_NormW32(energy) - 16;
+        // We want background_noise_.energy() / energy in Q14.
+        int32_t bgn_energy = WEBRTC_SPL_SHIFT_W32(
+            background_noise_.Energy(channel_ix), scaling + 14);
+        int16_t energy_scaled =
+            static_cast<int16_t>(WEBRTC_SPL_SHIFT_W32(energy, scaling));
+        int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
+        local_mute_factor =
+            std::min(local_mute_factor, WebRtcSpl_SqrtFloor(ratio << 14));
+      }
+      mute_factor = std::max<int16_t>(mute_factor, local_mute_factor);
+      RTC_DCHECK_LE(mute_factor, 16384);
+      RTC_DCHECK_GE(mute_factor, 0);
+
+      // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14),
+      // or as fast as it takes to come back to full gain within the frame
+      // length.
+      const int back_to_fullscale_inc =
+          static_cast<int>((16384 - mute_factor) / length_per_channel);
+      const int increment = std::max(64 / fs_mult, back_to_fullscale_inc);
+      for (size_t i = 0; i < length_per_channel; i++) {
+        // Scale with mute factor.
+        RTC_DCHECK_LT(channel_ix, output->Channels());
+        RTC_DCHECK_LT(i, output->Size());
+        int32_t scaled_signal = (*output)[channel_ix][i] * mute_factor;
+        // Shift 14 with proper rounding.
+        (*output)[channel_ix][i] =
+            static_cast<int16_t>((scaled_signal + 8192) >> 14);
+        // Increase mute_factor towards 16384.
+        mute_factor =
+            static_cast<int16_t>(std::min(mute_factor + increment, 16384));
+      }
+
+      // Interpolate the expanded data into the new vector.
+      // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
+      size_t win_length = samples_per_ms_;
+      int16_t win_slope_Q14 = default_win_slope_Q14_;
+      RTC_DCHECK_LT(channel_ix, output->Channels());
+      if (win_length > output->Size()) {
+        win_length = output->Size();
+        win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
+      }
+      int16_t win_up_Q14 = 0;
+      for (size_t i = 0; i < win_length; i++) {
+        win_up_Q14 += win_slope_Q14;
+        (*output)[channel_ix][i] =
+            (win_up_Q14 * (*output)[channel_ix][i] +
+             ((1 << 14) - win_up_Q14) * expanded[channel_ix][i] + (1 << 13)) >>
+            14;
+      }
+      RTC_DCHECK_GT(win_up_Q14,
+                    (1 << 14) - 32);  // Worst case rouding is a length of 34
+    }
+  } else if (last_mode == NetEq::Mode::kRfc3389Cng) {
+    RTC_DCHECK_EQ(output->Channels(), 1);  // Not adapted for multi-channel yet.
+    static const size_t kCngLength = 48;
+    RTC_DCHECK_LE(8 * fs_mult, kCngLength);
+    int16_t cng_output[kCngLength];
+    ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
+
+    if (cng_decoder) {
+      // Generate long enough for 48kHz.
+      if (!cng_decoder->Generate(cng_output, false)) {
+        // Error returned; set return vector to all zeros.
+        memset(cng_output, 0, sizeof(cng_output));
+      }
+    } else {
+      // If no CNG instance is defined, just copy from the decoded data.
+      // (This will result in interpolating the decoded with itself.)
+      (*output)[0].CopyTo(fs_mult * 8, 0, cng_output);
+    }
+    // Interpolate the CNG into the new vector.
+    // (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
+    size_t win_length = samples_per_ms_;
+    int16_t win_slope_Q14 = default_win_slope_Q14_;
+    if (win_length > kCngLength) {
+      win_length = kCngLength;
+      win_slope_Q14 = (1 << 14) / static_cast<int16_t>(win_length);
+    }
+    int16_t win_up_Q14 = 0;
+    for (size_t i = 0; i < win_length; i++) {
+      win_up_Q14 += win_slope_Q14;
+      (*output)[0][i] =
+          (win_up_Q14 * (*output)[0][i] +
+           ((1 << 14) - win_up_Q14) * cng_output[i] + (1 << 13)) >>
+          14;
+    }
+    RTC_DCHECK_GT(win_up_Q14,
+                  (1 << 14) - 32);  // Worst case rouding is a length of 34
+  }
+
+  return static_cast<int>(length);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/normal.h b/third_party/libwebrtc/modules/audio_coding/neteq/normal.h
new file mode 100644
index 0000000000..772293b605
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/normal.h
@@ -0,0 +1,76 @@
+/*
+ *  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_NETEQ_NORMAL_H_
+#define MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
+
+#include <stdint.h>
+#include <string.h>  // Access to size_t.
+
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class AudioMultiVector;
+class BackgroundNoise;
+class DecoderDatabase;
+class Expand;
+
+// This class provides the "Normal" DSP operation, that is performed when
+// there is no data loss, no need to stretch the timing of the signal, and
+// no other "special circumstances" are at hand.
+class Normal {
+ public:
+  Normal(int fs_hz,
+         DecoderDatabase* decoder_database,
+         const BackgroundNoise& background_noise,
+         Expand* expand,
+         StatisticsCalculator* statistics)
+      : fs_hz_(fs_hz),
+        decoder_database_(decoder_database),
+        background_noise_(background_noise),
+        expand_(expand),
+        samples_per_ms_(rtc::CheckedDivExact(fs_hz_, 1000)),
+        default_win_slope_Q14_(
+            rtc::dchecked_cast<uint16_t>((1 << 14) / samples_per_ms_)),
+        statistics_(statistics) {}
+
+  virtual ~Normal() {}
+
+  Normal(const Normal&) = delete;
+  Normal& operator=(const Normal&) = delete;
+
+  // Performs the "Normal" operation. The decoder data is supplied in `input`,
+  // having `length` samples in total for all channels (interleaved). The
+  // result is written to `output`. The number of channels allocated in
+  // `output` defines the number of channels that will be used when
+  // de-interleaving `input`. `last_mode` contains the mode used in the previous
+  // GetAudio call (i.e., not the current one).
+  int Process(const int16_t* input,
+              size_t length,
+              NetEq::Mode last_mode,
+              AudioMultiVector* output);
+
+ private:
+  int fs_hz_;
+  DecoderDatabase* decoder_database_;
+  const BackgroundNoise& background_noise_;
+  Expand* expand_;
+  const size_t samples_per_ms_;
+  const int16_t default_win_slope_Q14_;
+  StatisticsCalculator* const statistics_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/normal_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/normal_unittest.cc
new file mode 100644
index 0000000000..4554d79576
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/normal_unittest.cc
@@ -0,0 +1,147 @@
+/*
+ *  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.
+ */
+
+// Unit tests for Normal class.
+
+#include "modules/audio_coding/neteq/normal.h"
+
+#include <memory>
+#include <vector>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/mock/mock_expand.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+
+namespace webrtc {
+
+namespace {
+
+int ExpandProcess120ms(AudioMultiVector* output) {
+  AudioMultiVector dummy_audio(1, 11520u);
+  dummy_audio.CopyTo(output);
+  return 0;
+}
+
+}  // namespace
+
+TEST(Normal, CreateAndDestroy) {
+  MockDecoderDatabase db;
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels);
+  Normal normal(fs, &db, bgn, &expand, &statistics);
+  EXPECT_CALL(db, Die());  // Called when `db` goes out of scope.
+}
+
+TEST(Normal, AvoidDivideByZero) {
+  MockDecoderDatabase db;
+  int fs = 8000;
+  size_t channels = 1;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(1, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
+                    channels);
+  Normal normal(fs, &db, bgn, &expand, &statistics);
+
+  int16_t input[1000] = {0};
+  AudioMultiVector output(channels);
+
+  // Zero input length.
+  EXPECT_EQ(0, normal.Process(input, 0, NetEq::Mode::kExpand, &output));
+  EXPECT_EQ(0u, output.Size());
+
+  // Try to make energy_length >> scaling = 0;
+  EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
+  EXPECT_CALL(expand, Process(_));
+  EXPECT_CALL(expand, Reset());
+  // If input_size_samples < 64, then energy_length in Normal::Process() will
+  // be equal to input_size_samples. Since the input is all zeros, decoded_max
+  // will be zero, and scaling will be >= 6. Thus, energy_length >> scaling = 0,
+  // and using this as a denominator would lead to problems.
+  int input_size_samples = 63;
+  EXPECT_EQ(input_size_samples, normal.Process(input, input_size_samples,
+                                               NetEq::Mode::kExpand, &output));
+
+  EXPECT_CALL(db, Die());      // Called when `db` goes out of scope.
+  EXPECT_CALL(expand, Die());  // Called when `expand` goes out of scope.
+}
+
+TEST(Normal, InputLengthAndChannelsDoNotMatch) {
+  MockDecoderDatabase db;
+  int fs = 8000;
+  size_t channels = 2;
+  BackgroundNoise bgn(channels);
+  SyncBuffer sync_buffer(channels, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs,
+                    channels);
+  Normal normal(fs, &db, bgn, &expand, &statistics);
+
+  int16_t input[1000] = {0};
+  AudioMultiVector output(channels);
+
+  // Let the number of samples be one sample less than 80 samples per channel.
+  size_t input_len = 80 * channels - 1;
+  EXPECT_EQ(0, normal.Process(input, input_len, NetEq::Mode::kExpand, &output));
+  EXPECT_EQ(0u, output.Size());
+
+  EXPECT_CALL(db, Die());      // Called when `db` goes out of scope.
+  EXPECT_CALL(expand, Die());  // Called when `expand` goes out of scope.
+}
+
+TEST(Normal, LastModeExpand120msPacket) {
+  MockDecoderDatabase db;
+  const int kFs = 48000;
+  const size_t kPacketsizeBytes = 11520u;
+  const size_t kChannels = 1;
+  BackgroundNoise bgn(kChannels);
+  SyncBuffer sync_buffer(kChannels, 1000);
+  RandomVector random_vector;
+  StatisticsCalculator statistics;
+  MockExpand expand(&bgn, &sync_buffer, &random_vector, &statistics, kFs,
+                    kChannels);
+  Normal normal(kFs, &db, bgn, &expand, &statistics);
+
+  int16_t input[kPacketsizeBytes] = {0};
+  AudioMultiVector output(kChannels);
+
+  EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
+  EXPECT_CALL(expand, Process(_)).WillOnce(Invoke(ExpandProcess120ms));
+  EXPECT_CALL(expand, Reset());
+  EXPECT_EQ(
+      static_cast<int>(kPacketsizeBytes),
+      normal.Process(input, kPacketsizeBytes, NetEq::Mode::kExpand, &output));
+
+  EXPECT_EQ(kPacketsizeBytes, output.Size());
+
+  EXPECT_CALL(db, Die());      // Called when `db` goes out of scope.
+  EXPECT_CALL(expand, Die());  // Called when `expand` goes out of scope.
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet.cc
new file mode 100644
index 0000000000..333f161229
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet.cc
@@ -0,0 +1,36 @@
+/*
+ *  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/neteq/packet.h"
+
+namespace webrtc {
+
+Packet::Packet() = default;
+Packet::Packet(Packet&& b) = default;
+
+Packet::~Packet() = default;
+
+Packet& Packet::operator=(Packet&& b) = default;
+
+Packet Packet::Clone() const {
+  RTC_CHECK(!frame);
+
+  Packet clone;
+  clone.timestamp = timestamp;
+  clone.sequence_number = sequence_number;
+  clone.payload_type = payload_type;
+  clone.payload.SetData(payload.data(), payload.size());
+  clone.priority = priority;
+  clone.packet_info = packet_info;
+
+  return clone;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet.h
new file mode 100644
index 0000000000..0c6f204edb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet.h
@@ -0,0 +1,128 @@
+/*
+ *  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_NETEQ_PACKET_H_
+#define MODULES_AUDIO_CODING_NETEQ_PACKET_H_
+
+#include <stdint.h>
+
+#include <list>
+#include <memory>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/tick_timer.h"
+#include "api/rtp_packet_info.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Struct for holding RTP packets.
+struct Packet {
+  struct Priority {
+    Priority() : codec_level(0), red_level(0) {}
+    Priority(int codec_level, int red_level)
+        : codec_level(codec_level), red_level(red_level) {
+      CheckInvariant();
+    }
+
+    int codec_level;
+    int red_level;
+
+    // Priorities are sorted low-to-high, first on the level the codec
+    // prioritizes it, then on the level of RED packet it is; i.e. if it is a
+    // primary or secondary payload of a RED packet. For example: with Opus, an
+    // Fec packet (which the decoder prioritizes lower than a regular packet)
+    // will not be used if there is _any_ RED payload for the same
+    // timeframe. The highest priority packet will have levels {0, 0}. Negative
+    // priorities are not allowed.
+    bool operator<(const Priority& b) const {
+      CheckInvariant();
+      b.CheckInvariant();
+      if (codec_level == b.codec_level)
+        return red_level < b.red_level;
+
+      return codec_level < b.codec_level;
+    }
+    bool operator==(const Priority& b) const {
+      CheckInvariant();
+      b.CheckInvariant();
+      return codec_level == b.codec_level && red_level == b.red_level;
+    }
+    bool operator!=(const Priority& b) const { return !(*this == b); }
+    bool operator>(const Priority& b) const { return b < *this; }
+    bool operator<=(const Priority& b) const { return !(b > *this); }
+    bool operator>=(const Priority& b) const { return !(b < *this); }
+
+   private:
+    void CheckInvariant() const {
+      RTC_DCHECK_GE(codec_level, 0);
+      RTC_DCHECK_GE(red_level, 0);
+    }
+  };
+
+  uint32_t timestamp;
+  uint16_t sequence_number;
+  uint8_t payload_type;
+  // Datagram excluding RTP header and header extension.
+  rtc::Buffer payload;
+  Priority priority;
+  RtpPacketInfo packet_info;
+  std::unique_ptr<TickTimer::Stopwatch> waiting_time;
+  std::unique_ptr<AudioDecoder::EncodedAudioFrame> frame;
+
+  Packet();
+  Packet(Packet&& b);
+  ~Packet();
+
+  // Packets should generally be moved around but sometimes it's useful to make
+  // a copy, for example for testing purposes. NOTE: Will only work for
+  // un-parsed packets, i.e. `frame` must be unset. The payload will, however,
+  // be copied. `waiting_time` will also not be copied.
+  Packet Clone() const;
+
+  Packet& operator=(Packet&& b);
+
+  // Comparison operators. Establish a packet ordering based on (1) timestamp,
+  // (2) sequence number and (3) redundancy.
+  // Timestamp and sequence numbers are compared taking wrap-around into
+  // account. For two packets with the same sequence number and timestamp a
+  // primary payload is considered "smaller" than a secondary.
+  bool operator==(const Packet& rhs) const {
+    return (this->timestamp == rhs.timestamp &&
+            this->sequence_number == rhs.sequence_number &&
+            this->priority == rhs.priority);
+  }
+  bool operator!=(const Packet& rhs) const { return !operator==(rhs); }
+  bool operator<(const Packet& rhs) const {
+    if (this->timestamp == rhs.timestamp) {
+      if (this->sequence_number == rhs.sequence_number) {
+        // Timestamp and sequence numbers are identical - deem the left hand
+        // side to be "smaller" (i.e., "earlier") if it has higher priority.
+        return this->priority < rhs.priority;
+      }
+      return (static_cast<uint16_t>(rhs.sequence_number -
+                                    this->sequence_number) < 0xFFFF / 2);
+    }
+    return (static_cast<uint32_t>(rhs.timestamp - this->timestamp) <
+            0xFFFFFFFF / 2);
+  }
+  bool operator>(const Packet& rhs) const { return rhs.operator<(*this); }
+  bool operator<=(const Packet& rhs) const { return !operator>(rhs); }
+  bool operator>=(const Packet& rhs) const { return !operator<(rhs); }
+
+  bool empty() const { return !frame && payload.empty(); }
+};
+
+// A list of packets.
+typedef std::list<Packet> PacketList;
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_PACKET_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc
new file mode 100644
index 0000000000..2077383f76
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can 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/neteq/packet_arrival_history.h"
+
+#include <algorithm>
+
+#include "api/neteq/tick_timer.h"
+
+namespace webrtc {
+
+PacketArrivalHistory::PacketArrivalHistory(int window_size_ms)
+    : window_size_ms_(window_size_ms) {}
+
+void PacketArrivalHistory::Insert(uint32_t rtp_timestamp,
+                                  int64_t arrival_time_ms) {
+  RTC_DCHECK(sample_rate_khz_ > 0);
+  int64_t unwrapped_rtp_timestamp = timestamp_unwrapper_.Unwrap(rtp_timestamp);
+  if (!newest_rtp_timestamp_ ||
+      unwrapped_rtp_timestamp > *newest_rtp_timestamp_) {
+    newest_rtp_timestamp_ = unwrapped_rtp_timestamp;
+  }
+  history_.emplace_back(unwrapped_rtp_timestamp / sample_rate_khz_,
+                        arrival_time_ms);
+  MaybeUpdateCachedArrivals(history_.back());
+  while (history_.front().rtp_timestamp_ms + window_size_ms_ <
+         unwrapped_rtp_timestamp / sample_rate_khz_) {
+    if (&history_.front() == min_packet_arrival_) {
+      min_packet_arrival_ = nullptr;
+    }
+    if (&history_.front() == max_packet_arrival_) {
+      max_packet_arrival_ = nullptr;
+    }
+    history_.pop_front();
+  }
+  if (!min_packet_arrival_ || !max_packet_arrival_) {
+    for (const PacketArrival& packet : history_) {
+      MaybeUpdateCachedArrivals(packet);
+    }
+  }
+}
+
+void PacketArrivalHistory::MaybeUpdateCachedArrivals(
+    const PacketArrival& packet_arrival) {
+  if (!min_packet_arrival_ || packet_arrival <= *min_packet_arrival_) {
+    min_packet_arrival_ = &packet_arrival;
+  }
+  if (!max_packet_arrival_ || packet_arrival >= *max_packet_arrival_) {
+    max_packet_arrival_ = &packet_arrival;
+  }
+}
+
+void PacketArrivalHistory::Reset() {
+  history_.clear();
+  min_packet_arrival_ = nullptr;
+  max_packet_arrival_ = nullptr;
+  timestamp_unwrapper_.Reset();
+  newest_rtp_timestamp_ = absl::nullopt;
+}
+
+int PacketArrivalHistory::GetDelayMs(uint32_t rtp_timestamp,
+                                     int64_t time_ms) const {
+  RTC_DCHECK(sample_rate_khz_ > 0);
+  int64_t unwrapped_rtp_timestamp_ms =
+      timestamp_unwrapper_.PeekUnwrap(rtp_timestamp) / sample_rate_khz_;
+  PacketArrival packet(unwrapped_rtp_timestamp_ms, time_ms);
+  return GetPacketArrivalDelayMs(packet);
+}
+
+int PacketArrivalHistory::GetMaxDelayMs() const {
+  if (!max_packet_arrival_) {
+    return 0;
+  }
+  return GetPacketArrivalDelayMs(*max_packet_arrival_);
+}
+
+bool PacketArrivalHistory::IsNewestRtpTimestamp(uint32_t rtp_timestamp) const {
+  if (!newest_rtp_timestamp_) {
+    return false;
+  }
+  int64_t unwrapped_rtp_timestamp =
+      timestamp_unwrapper_.PeekUnwrap(rtp_timestamp);
+  return unwrapped_rtp_timestamp == *newest_rtp_timestamp_;
+}
+
+int PacketArrivalHistory::GetPacketArrivalDelayMs(
+    const PacketArrival& packet_arrival) const {
+  if (!min_packet_arrival_) {
+    return 0;
+  }
+  return std::max(static_cast<int>(packet_arrival.arrival_time_ms -
+                                   min_packet_arrival_->arrival_time_ms -
+                                   (packet_arrival.rtp_timestamp_ms -
+                                    min_packet_arrival_->rtp_timestamp_ms)),
+                  0);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h
new file mode 100644
index 0000000000..cad362b469
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can 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_NETEQ_PACKET_ARRIVAL_HISTORY_H_
+#define MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
+
+#include <cstdint>
+#include <deque>
+
+#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+
+// Stores timing information about previously received packets.
+// The history has a fixed window size beyond which old data is automatically
+// pruned.
+class PacketArrivalHistory {
+ public:
+  explicit PacketArrivalHistory(int window_size_ms);
+
+  // Insert packet with `rtp_timestamp` and `arrival_time_ms` into the history.
+  void Insert(uint32_t rtp_timestamp, int64_t arrival_time_ms);
+
+  // The delay for `rtp_timestamp` at `time_ms` is calculated as
+  // `(time_ms - p.arrival_time_ms) - (rtp_timestamp - p.rtp_timestamp)`
+  // where `p` is chosen as the packet arrival in the history that maximizes the
+  // delay.
+  int GetDelayMs(uint32_t rtp_timestamp, int64_t time_ms) const;
+
+  // Get the maximum packet arrival delay observed in the history.
+  int GetMaxDelayMs() const;
+
+  bool IsNewestRtpTimestamp(uint32_t rtp_timestamp) const;
+
+  void Reset();
+
+  void set_sample_rate(int sample_rate) {
+    sample_rate_khz_ = sample_rate / 1000;
+  }
+
+  size_t size() const { return history_.size(); }
+
+ private:
+  struct PacketArrival {
+    PacketArrival(int64_t rtp_timestamp_ms, int64_t arrival_time_ms)
+        : rtp_timestamp_ms(rtp_timestamp_ms),
+          arrival_time_ms(arrival_time_ms) {}
+    int64_t rtp_timestamp_ms;
+    int64_t arrival_time_ms;
+    bool operator<=(const PacketArrival& other) const {
+      return arrival_time_ms - rtp_timestamp_ms <=
+             other.arrival_time_ms - other.rtp_timestamp_ms;
+    }
+    bool operator>=(const PacketArrival& other) const {
+      return arrival_time_ms - rtp_timestamp_ms >=
+             other.arrival_time_ms - other.rtp_timestamp_ms;
+    }
+  };
+  std::deque<PacketArrival> history_;
+  int GetPacketArrivalDelayMs(const PacketArrival& packet_arrival) const;
+  // Updates `min_packet_arrival_` and `max_packet_arrival_`.
+  void MaybeUpdateCachedArrivals(const PacketArrival& packet);
+  const PacketArrival* min_packet_arrival_ = nullptr;
+  const PacketArrival* max_packet_arrival_ = nullptr;
+  const int window_size_ms_;
+  RtpTimestampUnwrapper timestamp_unwrapper_;
+  absl::optional<int64_t> newest_rtp_timestamp_;
+  int sample_rate_khz_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history_unittest.cc
new file mode 100644
index 0000000000..539a318fe1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history_unittest.cc
@@ -0,0 +1,141 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can 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/neteq/packet_arrival_history.h"
+
+#include <cstdint>
+#include <limits>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kFs = 8000;
+constexpr int kFsKhz = kFs / 1000;
+constexpr int kFrameSizeMs = 20;
+constexpr int kWindowSizeMs = 1000;
+
+class PacketArrivalHistoryTest : public testing::Test {
+ public:
+  PacketArrivalHistoryTest() : history_(kWindowSizeMs) {
+    history_.set_sample_rate(kFs);
+  }
+  void IncrementTime(int delta_ms) { time_ms_ += delta_ms; }
+  int InsertPacketAndGetDelay(int timestamp_delta_ms) {
+    uint32_t timestamp = timestamp_ + timestamp_delta_ms * kFsKhz;
+    if (timestamp_delta_ms > 0) {
+      timestamp_ = timestamp;
+    }
+    history_.Insert(timestamp, time_ms_);
+    EXPECT_EQ(history_.IsNewestRtpTimestamp(timestamp),
+              timestamp_delta_ms >= 0);
+    return history_.GetDelayMs(timestamp, time_ms_);
+  }
+
+ protected:
+  int64_t time_ms_ = 0;
+  PacketArrivalHistory history_;
+  uint32_t timestamp_ = 0x12345678;
+};
+
+TEST_F(PacketArrivalHistoryTest, RelativeArrivalDelay) {
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  IncrementTime(kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 20);
+
+  // Reordered packet.
+  EXPECT_EQ(InsertPacketAndGetDelay(-2 * kFrameSizeMs), 60);
+
+  IncrementTime(2 * kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 40);
+
+  // Move reference packet forward.
+  EXPECT_EQ(InsertPacketAndGetDelay(4 * kFrameSizeMs), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 20);
+
+  // Earlier packet is now more delayed due to the new reference packet.
+  EXPECT_EQ(history_.GetMaxDelayMs(), 100);
+}
+
+TEST_F(PacketArrivalHistoryTest, ReorderedPackets) {
+  // Insert first packet.
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  // Insert reordered packet.
+  EXPECT_EQ(InsertPacketAndGetDelay(-80), 80);
+
+  // Insert another reordered packet.
+  EXPECT_EQ(InsertPacketAndGetDelay(-kFrameSizeMs), 20);
+
+  // Insert the next packet in order and verify that the relative delay is
+  // estimated based on the first inserted packet.
+  IncrementTime(4 * kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 60);
+
+  EXPECT_EQ(history_.GetMaxDelayMs(), 80);
+}
+
+TEST_F(PacketArrivalHistoryTest, MaxHistorySize) {
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 20);
+  EXPECT_EQ(history_.GetMaxDelayMs(), 20);
+
+  // Insert next packet with a timestamp difference larger than maximum history
+  // size. This removes the previously inserted packet from the history.
+  IncrementTime(kWindowSizeMs + kFrameSizeMs);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs + kWindowSizeMs), 0);
+  EXPECT_EQ(history_.GetMaxDelayMs(), 0);
+}
+
+TEST_F(PacketArrivalHistoryTest, TimestampWraparound) {
+  timestamp_ = std::numeric_limits<uint32_t>::max();
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  // Insert timestamp that will wrap around.
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), kFrameSizeMs);
+
+  // Insert reordered packet before the wraparound.
+  EXPECT_EQ(InsertPacketAndGetDelay(-2 * kFrameSizeMs), 3 * kFrameSizeMs);
+
+  // Insert another in-order packet after the wraparound.
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 0);
+
+  EXPECT_EQ(history_.GetMaxDelayMs(), 3 * kFrameSizeMs);
+}
+
+TEST_F(PacketArrivalHistoryTest, TimestampWraparoundBackwards) {
+  timestamp_ = 0;
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+
+  IncrementTime(2 * kFrameSizeMs);
+  // Insert timestamp that will wrap around.
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), kFrameSizeMs);
+
+  // Insert reordered packet before the wraparound.
+  EXPECT_EQ(InsertPacketAndGetDelay(-2 * kFrameSizeMs), 3 * kFrameSizeMs);
+
+  // Insert another in-order packet after the wraparound.
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 0);
+
+  EXPECT_EQ(history_.GetMaxDelayMs(), 3 * kFrameSizeMs);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc
new file mode 100644
index 0000000000..f6b5a476c9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc
@@ -0,0 +1,405 @@
+/*
+ *  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 is the implementation of the PacketBuffer class. It is mostly based on
+// an STL list. The list is kept sorted at all times so that the next packet to
+// decode is at the beginning of the list.
+
+#include "modules/audio_coding/neteq/packet_buffer.h"
+
+#include <algorithm>
+#include <list>
+#include <memory>
+#include <type_traits>
+#include <utility>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+// Predicate used when inserting packets in the buffer list.
+// Operator() returns true when `packet` goes before `new_packet`.
+class NewTimestampIsLarger {
+ public:
+  explicit NewTimestampIsLarger(const Packet& new_packet)
+      : new_packet_(new_packet) {}
+  bool operator()(const Packet& packet) { return (new_packet_ >= packet); }
+
+ private:
+  const Packet& new_packet_;
+};
+
+// Returns true if both payload types are known to the decoder database, and
+// have the same sample rate.
+bool EqualSampleRates(uint8_t pt1,
+                      uint8_t pt2,
+                      const DecoderDatabase& decoder_database) {
+  auto* di1 = decoder_database.GetDecoderInfo(pt1);
+  auto* di2 = decoder_database.GetDecoderInfo(pt2);
+  return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz();
+}
+
+void LogPacketDiscarded(int codec_level, StatisticsCalculator* stats) {
+  RTC_CHECK(stats);
+  if (codec_level > 0) {
+    stats->SecondaryPacketsDiscarded(1);
+  } else {
+    stats->PacketsDiscarded(1);
+  }
+}
+
+absl::optional<SmartFlushingConfig> GetSmartflushingConfig() {
+  absl::optional<SmartFlushingConfig> result;
+  std::string field_trial_string =
+      field_trial::FindFullName("WebRTC-Audio-NetEqSmartFlushing");
+  result = SmartFlushingConfig();
+  bool enabled = false;
+  auto parser = StructParametersParser::Create(
+      "enabled", &enabled, "target_level_threshold_ms",
+      &result->target_level_threshold_ms, "target_level_multiplier",
+      &result->target_level_multiplier);
+  parser->Parse(field_trial_string);
+  if (!enabled) {
+    return absl::nullopt;
+  }
+  RTC_LOG(LS_INFO) << "Using smart flushing, target_level_threshold_ms: "
+                   << result->target_level_threshold_ms
+                   << ", target_level_multiplier: "
+                   << result->target_level_multiplier;
+  return result;
+}
+
+}  // namespace
+
+PacketBuffer::PacketBuffer(size_t max_number_of_packets,
+                           const TickTimer* tick_timer)
+    : smart_flushing_config_(GetSmartflushingConfig()),
+      max_number_of_packets_(max_number_of_packets),
+      tick_timer_(tick_timer) {}
+
+// Destructor. All packets in the buffer will be destroyed.
+PacketBuffer::~PacketBuffer() {
+  buffer_.clear();
+}
+
+// Flush the buffer. All packets in the buffer will be destroyed.
+void PacketBuffer::Flush(StatisticsCalculator* stats) {
+  for (auto& p : buffer_) {
+    LogPacketDiscarded(p.priority.codec_level, stats);
+  }
+  buffer_.clear();
+  stats->FlushedPacketBuffer();
+}
+
+void PacketBuffer::PartialFlush(int target_level_ms,
+                                size_t sample_rate,
+                                size_t last_decoded_length,
+                                StatisticsCalculator* stats) {
+  // Make sure that at least half the packet buffer capacity will be available
+  // after the flush. This is done to avoid getting stuck if the target level is
+  // very high.
+  int target_level_samples =
+      std::min(target_level_ms * sample_rate / 1000,
+               max_number_of_packets_ * last_decoded_length / 2);
+  // We should avoid flushing to very low levels.
+  target_level_samples = std::max(
+      target_level_samples, smart_flushing_config_->target_level_threshold_ms);
+  while (GetSpanSamples(last_decoded_length, sample_rate, true) >
+             static_cast<size_t>(target_level_samples) ||
+         buffer_.size() > max_number_of_packets_ / 2) {
+    LogPacketDiscarded(PeekNextPacket()->priority.codec_level, stats);
+    buffer_.pop_front();
+  }
+}
+
+bool PacketBuffer::Empty() const {
+  return buffer_.empty();
+}
+
+int PacketBuffer::InsertPacket(Packet&& packet,
+                               StatisticsCalculator* stats,
+                               size_t last_decoded_length,
+                               size_t sample_rate,
+                               int target_level_ms,
+                               const DecoderDatabase& decoder_database) {
+  if (packet.empty()) {
+    RTC_LOG(LS_WARNING) << "InsertPacket invalid packet";
+    return kInvalidPacket;
+  }
+
+  RTC_DCHECK_GE(packet.priority.codec_level, 0);
+  RTC_DCHECK_GE(packet.priority.red_level, 0);
+
+  int return_val = kOK;
+
+  packet.waiting_time = tick_timer_->GetNewStopwatch();
+
+  // Perform a smart flush if the buffer size exceeds a multiple of the target
+  // level.
+  const size_t span_threshold =
+      smart_flushing_config_
+          ? smart_flushing_config_->target_level_multiplier *
+                std::max(smart_flushing_config_->target_level_threshold_ms,
+                         target_level_ms) *
+                sample_rate / 1000
+          : 0;
+  const bool smart_flush =
+      smart_flushing_config_.has_value() &&
+      GetSpanSamples(last_decoded_length, sample_rate, true) >= span_threshold;
+  if (buffer_.size() >= max_number_of_packets_ || smart_flush) {
+    size_t buffer_size_before_flush = buffer_.size();
+    if (smart_flushing_config_.has_value()) {
+      // Flush down to the target level.
+      PartialFlush(target_level_ms, sample_rate, last_decoded_length, stats);
+      return_val = kPartialFlush;
+    } else {
+      // Buffer is full.
+      Flush(stats);
+      return_val = kFlushed;
+    }
+    RTC_LOG(LS_WARNING) << "Packet buffer flushed, "
+                        << (buffer_size_before_flush - buffer_.size())
+                        << " packets discarded.";
+  }
+
+  // Get an iterator pointing to the place in the buffer where the new packet
+  // should be inserted. The list is searched from the back, since the most
+  // likely case is that the new packet should be near the end of the list.
+  PacketList::reverse_iterator rit = std::find_if(
+      buffer_.rbegin(), buffer_.rend(), NewTimestampIsLarger(packet));
+
+  // The new packet is to be inserted to the right of `rit`. If it has the same
+  // timestamp as `rit`, which has a higher priority, do not insert the new
+  // packet to list.
+  if (rit != buffer_.rend() && packet.timestamp == rit->timestamp) {
+    LogPacketDiscarded(packet.priority.codec_level, stats);
+    return return_val;
+  }
+
+  // The new packet is to be inserted to the left of `it`. If it has the same
+  // timestamp as `it`, which has a lower priority, replace `it` with the new
+  // packet.
+  PacketList::iterator it = rit.base();
+  if (it != buffer_.end() && packet.timestamp == it->timestamp) {
+    LogPacketDiscarded(it->priority.codec_level, stats);
+    it = buffer_.erase(it);
+  }
+  buffer_.insert(it, std::move(packet));  // Insert the packet at that position.
+
+  return return_val;
+}
+
+int PacketBuffer::InsertPacketList(
+    PacketList* packet_list,
+    const DecoderDatabase& decoder_database,
+    absl::optional<uint8_t>* current_rtp_payload_type,
+    absl::optional<uint8_t>* current_cng_rtp_payload_type,
+    StatisticsCalculator* stats,
+    size_t last_decoded_length,
+    size_t sample_rate,
+    int target_level_ms) {
+  RTC_DCHECK(stats);
+  bool flushed = false;
+  for (auto& packet : *packet_list) {
+    if (decoder_database.IsComfortNoise(packet.payload_type)) {
+      if (*current_cng_rtp_payload_type &&
+          **current_cng_rtp_payload_type != packet.payload_type) {
+        // New CNG payload type implies new codec type.
+        *current_rtp_payload_type = absl::nullopt;
+        Flush(stats);
+        flushed = true;
+      }
+      *current_cng_rtp_payload_type = packet.payload_type;
+    } else if (!decoder_database.IsDtmf(packet.payload_type)) {
+      // This must be speech.
+      if ((*current_rtp_payload_type &&
+           **current_rtp_payload_type != packet.payload_type) ||
+          (*current_cng_rtp_payload_type &&
+           !EqualSampleRates(packet.payload_type,
+                             **current_cng_rtp_payload_type,
+                             decoder_database))) {
+        *current_cng_rtp_payload_type = absl::nullopt;
+        Flush(stats);
+        flushed = true;
+      }
+      *current_rtp_payload_type = packet.payload_type;
+    }
+    int return_val =
+        InsertPacket(std::move(packet), stats, last_decoded_length, sample_rate,
+                     target_level_ms, decoder_database);
+    if (return_val == kFlushed) {
+      // The buffer flushed, but this is not an error. We can still continue.
+      flushed = true;
+    } else if (return_val != kOK) {
+      // An error occurred. Delete remaining packets in list and return.
+      packet_list->clear();
+      return return_val;
+    }
+  }
+  packet_list->clear();
+  return flushed ? kFlushed : kOK;
+}
+
+int PacketBuffer::NextTimestamp(uint32_t* next_timestamp) const {
+  if (Empty()) {
+    return kBufferEmpty;
+  }
+  if (!next_timestamp) {
+    return kInvalidPointer;
+  }
+  *next_timestamp = buffer_.front().timestamp;
+  return kOK;
+}
+
+int PacketBuffer::NextHigherTimestamp(uint32_t timestamp,
+                                      uint32_t* next_timestamp) const {
+  if (Empty()) {
+    return kBufferEmpty;
+  }
+  if (!next_timestamp) {
+    return kInvalidPointer;
+  }
+  PacketList::const_iterator it;
+  for (it = buffer_.begin(); it != buffer_.end(); ++it) {
+    if (it->timestamp >= timestamp) {
+      // Found a packet matching the search.
+      *next_timestamp = it->timestamp;
+      return kOK;
+    }
+  }
+  return kNotFound;
+}
+
+const Packet* PacketBuffer::PeekNextPacket() const {
+  return buffer_.empty() ? nullptr : &buffer_.front();
+}
+
+absl::optional<Packet> PacketBuffer::GetNextPacket() {
+  if (Empty()) {
+    // Buffer is empty.
+    return absl::nullopt;
+  }
+
+  absl::optional<Packet> packet(std::move(buffer_.front()));
+  // Assert that the packet sanity checks in InsertPacket method works.
+  RTC_DCHECK(!packet->empty());
+  buffer_.pop_front();
+
+  return packet;
+}
+
+int PacketBuffer::DiscardNextPacket(StatisticsCalculator* stats) {
+  if (Empty()) {
+    return kBufferEmpty;
+  }
+  // Assert that the packet sanity checks in InsertPacket method works.
+  const Packet& packet = buffer_.front();
+  RTC_DCHECK(!packet.empty());
+  LogPacketDiscarded(packet.priority.codec_level, stats);
+  buffer_.pop_front();
+  return kOK;
+}
+
+void PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit,
+                                     uint32_t horizon_samples,
+                                     StatisticsCalculator* stats) {
+  buffer_.remove_if([timestamp_limit, horizon_samples, stats](const Packet& p) {
+    if (timestamp_limit == p.timestamp ||
+        !IsObsoleteTimestamp(p.timestamp, timestamp_limit, horizon_samples)) {
+      return false;
+    }
+    LogPacketDiscarded(p.priority.codec_level, stats);
+    return true;
+  });
+}
+
+void PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit,
+                                        StatisticsCalculator* stats) {
+  DiscardOldPackets(timestamp_limit, 0, stats);
+}
+
+void PacketBuffer::DiscardPacketsWithPayloadType(uint8_t payload_type,
+                                                 StatisticsCalculator* stats) {
+  buffer_.remove_if([payload_type, stats](const Packet& p) {
+    if (p.payload_type != payload_type) {
+      return false;
+    }
+    LogPacketDiscarded(p.priority.codec_level, stats);
+    return true;
+  });
+}
+
+size_t PacketBuffer::NumPacketsInBuffer() const {
+  return buffer_.size();
+}
+
+size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const {
+  size_t num_samples = 0;
+  size_t last_duration = last_decoded_length;
+  for (const Packet& packet : buffer_) {
+    if (packet.frame) {
+      // TODO(hlundin): Verify that it's fine to count all packets and remove
+      // this check.
+      if (packet.priority != Packet::Priority(0, 0)) {
+        continue;
+      }
+      size_t duration = packet.frame->Duration();
+      if (duration > 0) {
+        last_duration = duration;  // Save the most up-to-date (valid) duration.
+      }
+    }
+    num_samples += last_duration;
+  }
+  return num_samples;
+}
+
+size_t PacketBuffer::GetSpanSamples(size_t last_decoded_length,
+                                    size_t sample_rate,
+                                    bool count_dtx_waiting_time) const {
+  if (buffer_.size() == 0) {
+    return 0;
+  }
+
+  size_t span = buffer_.back().timestamp - buffer_.front().timestamp;
+  if (buffer_.back().frame && buffer_.back().frame->Duration() > 0) {
+    size_t duration = buffer_.back().frame->Duration();
+    if (count_dtx_waiting_time && buffer_.back().frame->IsDtxPacket()) {
+      size_t waiting_time_samples = rtc::dchecked_cast<size_t>(
+          buffer_.back().waiting_time->ElapsedMs() * (sample_rate / 1000));
+      duration = std::max(duration, waiting_time_samples);
+    }
+    span += duration;
+  } else {
+    span += last_decoded_length;
+  }
+  return span;
+}
+
+bool PacketBuffer::ContainsDtxOrCngPacket(
+    const DecoderDatabase* decoder_database) const {
+  RTC_DCHECK(decoder_database);
+  for (const Packet& packet : buffer_) {
+    if ((packet.frame && packet.frame->IsDtxPacket()) ||
+        decoder_database->IsComfortNoise(packet.payload_type)) {
+      return true;
+    }
+  }
+  return false;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.h
new file mode 100644
index 0000000000..c6fb47ffbf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.h
@@ -0,0 +1,181 @@
+/*
+ *  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_NETEQ_PACKET_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "modules/include/module_common_types_public.h"  // IsNewerTimestamp
+
+namespace webrtc {
+
+class DecoderDatabase;
+class StatisticsCalculator;
+class TickTimer;
+struct SmartFlushingConfig {
+  // When calculating the flushing threshold, the maximum between the target
+  // level and this value is used.
+  int target_level_threshold_ms = 500;
+  // A smart flush is triggered when the packet buffer contains a multiple of
+  // the target level.
+  int target_level_multiplier = 3;
+};
+
+// This is the actual buffer holding the packets before decoding.
+class PacketBuffer {
+ public:
+  enum BufferReturnCodes {
+    kOK = 0,
+    kFlushed,
+    kPartialFlush,
+    kNotFound,
+    kBufferEmpty,
+    kInvalidPacket,
+    kInvalidPointer
+  };
+
+  // Constructor creates a buffer which can hold a maximum of
+  // `max_number_of_packets` packets.
+  PacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer);
+
+  // Deletes all packets in the buffer before destroying the buffer.
+  virtual ~PacketBuffer();
+
+  PacketBuffer(const PacketBuffer&) = delete;
+  PacketBuffer& operator=(const PacketBuffer&) = delete;
+
+  // Flushes the buffer and deletes all packets in it.
+  virtual void Flush(StatisticsCalculator* stats);
+
+  // Partial flush. Flush packets but leave some packets behind.
+  virtual void PartialFlush(int target_level_ms,
+                            size_t sample_rate,
+                            size_t last_decoded_length,
+                            StatisticsCalculator* stats);
+
+  // Returns true for an empty buffer.
+  virtual bool Empty() const;
+
+  // Inserts `packet` into the buffer. The buffer will take over ownership of
+  // the packet object.
+  // Returns PacketBuffer::kOK on success, PacketBuffer::kFlushed if the buffer
+  // was flushed due to overfilling.
+  virtual int InsertPacket(Packet&& packet,
+                           StatisticsCalculator* stats,
+                           size_t last_decoded_length,
+                           size_t sample_rate,
+                           int target_level_ms,
+                           const DecoderDatabase& decoder_database);
+
+  // Inserts a list of packets into the buffer. The buffer will take over
+  // ownership of the packet objects.
+  // Returns PacketBuffer::kOK if all packets were inserted successfully.
+  // If the buffer was flushed due to overfilling, only a subset of the list is
+  // inserted, and PacketBuffer::kFlushed is returned.
+  // The last three parameters are included for legacy compatibility.
+  // TODO(hlundin): Redesign to not use current_*_payload_type and
+  // decoder_database.
+  virtual int InsertPacketList(
+      PacketList* packet_list,
+      const DecoderDatabase& decoder_database,
+      absl::optional<uint8_t>* current_rtp_payload_type,
+      absl::optional<uint8_t>* current_cng_rtp_payload_type,
+      StatisticsCalculator* stats,
+      size_t last_decoded_length,
+      size_t sample_rate,
+      int target_level_ms);
+
+  // Gets the timestamp for the first packet in the buffer and writes it to the
+  // output variable `next_timestamp`.
+  // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
+  // PacketBuffer::kOK otherwise.
+  virtual int NextTimestamp(uint32_t* next_timestamp) const;
+
+  // Gets the timestamp for the first packet in the buffer with a timestamp no
+  // lower than the input limit `timestamp`. The result is written to the output
+  // variable `next_timestamp`.
+  // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
+  // PacketBuffer::kOK otherwise.
+  virtual int NextHigherTimestamp(uint32_t timestamp,
+                                  uint32_t* next_timestamp) const;
+
+  // Returns a (constant) pointer to the first packet in the buffer. Returns
+  // NULL if the buffer is empty.
+  virtual const Packet* PeekNextPacket() const;
+
+  // Extracts the first packet in the buffer and returns it.
+  // Returns an empty optional if the buffer is empty.
+  virtual absl::optional<Packet> GetNextPacket();
+
+  // Discards the first packet in the buffer. The packet is deleted.
+  // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
+  // PacketBuffer::kOK otherwise.
+  virtual int DiscardNextPacket(StatisticsCalculator* stats);
+
+  // Discards all packets that are (strictly) older than timestamp_limit,
+  // but newer than timestamp_limit - horizon_samples. Setting horizon_samples
+  // to zero implies that the horizon is set to half the timestamp range. That
+  // is, if a packet is more than 2^31 timestamps into the future compared with
+  // timestamp_limit (including wrap-around), it is considered old.
+  virtual void DiscardOldPackets(uint32_t timestamp_limit,
+                                 uint32_t horizon_samples,
+                                 StatisticsCalculator* stats);
+
+  // Discards all packets that are (strictly) older than timestamp_limit.
+  virtual void DiscardAllOldPackets(uint32_t timestamp_limit,
+                                    StatisticsCalculator* stats);
+
+  // Removes all packets with a specific payload type from the buffer.
+  virtual void DiscardPacketsWithPayloadType(uint8_t payload_type,
+                                             StatisticsCalculator* stats);
+
+  // Returns the number of packets in the buffer, including duplicates and
+  // redundant packets.
+  virtual size_t NumPacketsInBuffer() const;
+
+  // Returns the number of samples in the buffer, including samples carried in
+  // duplicate and redundant packets.
+  virtual size_t NumSamplesInBuffer(size_t last_decoded_length) const;
+
+  // Returns the total duration in samples that the packets in the buffer spans
+  // across.
+  virtual size_t GetSpanSamples(size_t last_decoded_length,
+                                size_t sample_rate,
+                                bool count_dtx_waiting_time) const;
+
+  // Returns true if the packet buffer contains any DTX or CNG packets.
+  virtual bool ContainsDtxOrCngPacket(
+      const DecoderDatabase* decoder_database) const;
+
+  // Static method returning true if `timestamp` is older than `timestamp_limit`
+  // but less than `horizon_samples` behind `timestamp_limit`. For instance,
+  // with timestamp_limit = 100 and horizon_samples = 10, a timestamp in the
+  // range (90, 100) is considered obsolete, and will yield true.
+  // Setting `horizon_samples` to 0 is the same as setting it to 2^31, i.e.,
+  // half the 32-bit timestamp range.
+  static bool IsObsoleteTimestamp(uint32_t timestamp,
+                                  uint32_t timestamp_limit,
+                                  uint32_t horizon_samples) {
+    return IsNewerTimestamp(timestamp_limit, timestamp) &&
+           (horizon_samples == 0 ||
+            IsNewerTimestamp(timestamp, timestamp_limit - horizon_samples));
+  }
+
+ private:
+  absl::optional<SmartFlushingConfig> smart_flushing_config_;
+  size_t max_number_of_packets_;
+  PacketList buffer_;
+  const TickTimer* tick_timer_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
new file mode 100644
index 0000000000..1a054daca3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -0,0 +1,989 @@
+/*
+ *  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.
+ */
+
+// Unit tests for PacketBuffer class.
+
+#include "modules/audio_coding/neteq/packet_buffer.h"
+
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::InSequence;
+using ::testing::MockFunction;
+using ::testing::Return;
+using ::testing::StrictMock;
+
+namespace {
+class MockEncodedAudioFrame : public webrtc::AudioDecoder::EncodedAudioFrame {
+ public:
+  MOCK_METHOD(size_t, Duration, (), (const, override));
+
+  MOCK_METHOD(bool, IsDtxPacket, (), (const, override));
+
+  MOCK_METHOD(absl::optional<DecodeResult>,
+              Decode,
+              (rtc::ArrayView<int16_t> decoded),
+              (const, override));
+};
+
+// Helper class to generate packets. Packets must be deleted by the user.
+class PacketGenerator {
+ public:
+  PacketGenerator(uint16_t seq_no, uint32_t ts, uint8_t pt, int frame_size);
+  virtual ~PacketGenerator() {}
+  void Reset(uint16_t seq_no, uint32_t ts, uint8_t pt, int frame_size);
+  webrtc::Packet NextPacket(
+      int payload_size_bytes,
+      std::unique_ptr<webrtc::AudioDecoder::EncodedAudioFrame> audio_frame);
+
+  uint16_t seq_no_;
+  uint32_t ts_;
+  uint8_t pt_;
+  int frame_size_;
+};
+
+PacketGenerator::PacketGenerator(uint16_t seq_no,
+                                 uint32_t ts,
+                                 uint8_t pt,
+                                 int frame_size) {
+  Reset(seq_no, ts, pt, frame_size);
+}
+
+void PacketGenerator::Reset(uint16_t seq_no,
+                            uint32_t ts,
+                            uint8_t pt,
+                            int frame_size) {
+  seq_no_ = seq_no;
+  ts_ = ts;
+  pt_ = pt;
+  frame_size_ = frame_size;
+}
+
+webrtc::Packet PacketGenerator::NextPacket(
+    int payload_size_bytes,
+    std::unique_ptr<webrtc::AudioDecoder::EncodedAudioFrame> audio_frame) {
+  webrtc::Packet packet;
+  packet.sequence_number = seq_no_;
+  packet.timestamp = ts_;
+  packet.payload_type = pt_;
+  packet.payload.SetSize(payload_size_bytes);
+  ++seq_no_;
+  ts_ += frame_size_;
+  packet.frame = std::move(audio_frame);
+  return packet;
+}
+
+struct PacketsToInsert {
+  uint16_t sequence_number;
+  uint32_t timestamp;
+  uint8_t payload_type;
+  bool primary;
+  // Order of this packet to appear upon extraction, after inserting a series
+  // of packets. A negative number means that it should have been discarded
+  // before extraction.
+  int extract_order;
+};
+
+}  // namespace
+
+namespace webrtc {
+
+// Start of test definitions.
+
+TEST(PacketBuffer, CreateAndDestroy) {
+  TickTimer tick_timer;
+  PacketBuffer* buffer = new PacketBuffer(10, &tick_timer);  // 10 packets.
+  EXPECT_TRUE(buffer->Empty());
+  delete buffer;
+}
+
+TEST(PacketBuffer, InsertPacket) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(17u, 4711u, 0, 10);
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  const int payload_len = 100;
+  const Packet packet = gen.NextPacket(payload_len, nullptr);
+  EXPECT_EQ(0, buffer.InsertPacket(/*packet=*/packet.Clone(),
+                                   /*stats=*/&mock_stats,
+                                   /*last_decoded_length=*/payload_len,
+                                   /*sample_rate=*/10000,
+                                   /*target_level_ms=*/60,
+                                   /*decoder_database=*/decoder_database));
+  uint32_t next_ts;
+  EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
+  EXPECT_EQ(4711u, next_ts);
+  EXPECT_FALSE(buffer.Empty());
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
+  const Packet* next_packet = buffer.PeekNextPacket();
+  EXPECT_EQ(packet, *next_packet);  // Compare contents.
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+
+  // Do not explicitly flush buffer or delete packet to test that it is deleted
+  // with the buffer. (Tested with Valgrind or similar tool.)
+}
+
+// Test to flush buffer.
+TEST(PacketBuffer, FlushBuffer) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  const int payload_len = 10;
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  // Insert 10 small packets; should be ok.
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(
+        PacketBuffer::kOK,
+        buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                            /*stats=*/&mock_stats,
+                            /*last_decoded_length=*/payload_len,
+                            /*sample_rate=*/1000,
+                            /*target_level_ms=*/60,
+                            /*decoder_database=*/decoder_database));
+  }
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+  EXPECT_FALSE(buffer.Empty());
+
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(10);
+  buffer.Flush(&mock_stats);
+  // Buffer should delete the payloads itself.
+  EXPECT_EQ(0u, buffer.NumPacketsInBuffer());
+  EXPECT_TRUE(buffer.Empty());
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test to fill the buffer over the limits, and verify that it flushes.
+TEST(PacketBuffer, OverfillBuffer) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  // Insert 10 small packets; should be ok.
+  const int payload_len = 10;
+  int i;
+  for (i = 0; i < 10; ++i) {
+    EXPECT_EQ(
+        PacketBuffer::kOK,
+        buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                            /*stats=*/&mock_stats,
+                            /*last_decoded_length=*/payload_len,
+                            /*sample_rate=*/1000,
+                            /*target_level_ms=*/60,
+                            /*decoder_database=*/decoder_database));
+  }
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+  uint32_t next_ts;
+  EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
+  EXPECT_EQ(0u, next_ts);  // Expect first inserted packet to be first in line.
+
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(10);
+  const Packet packet = gen.NextPacket(payload_len, nullptr);
+  // Insert 11th packet; should flush the buffer and insert it after flushing.
+  EXPECT_EQ(PacketBuffer::kFlushed,
+            buffer.InsertPacket(/*packet=*/packet.Clone(),
+                                /*stats=*/&mock_stats,
+                                /*last_decoded_length=*/payload_len,
+                                /*sample_rate=*/1000,
+                                /*target_level_ms=*/60,
+                                /*decoder_database=*/decoder_database));
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
+  // Expect last inserted packet to be first in line.
+  EXPECT_EQ(packet.timestamp, next_ts);
+
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test a partial buffer flush.
+TEST(PacketBuffer, PartialFlush) {
+  // Use a field trial to configure smart flushing.
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-NetEqSmartFlushing/enabled:true,"
+      "target_level_threshold_ms:0,target_level_multiplier:2/");
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  const int payload_len = 10;
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  // Insert 10 small packets; should be ok.
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(
+        PacketBuffer::kOK,
+        buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                            /*stats=*/&mock_stats,
+                            /*last_decoded_length=*/payload_len,
+                            /*sample_rate=*/1000,
+                            /*target_level_ms=*/100,
+                            /*decoder_database=*/decoder_database));
+  }
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+  EXPECT_FALSE(buffer.Empty());
+
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(7);
+  buffer.PartialFlush(/*target_level_ms=*/30,
+                      /*sample_rate=*/1000,
+                      /*last_decoded_length=*/payload_len,
+                      /*stats=*/&mock_stats);
+  // There should still be some packets left in the buffer.
+  EXPECT_EQ(3u, buffer.NumPacketsInBuffer());
+  EXPECT_FALSE(buffer.Empty());
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test to fill the buffer over the limits, and verify that the smart flush
+// functionality works as expected.
+TEST(PacketBuffer, SmartFlushOverfillBuffer) {
+  // Use a field trial to configure smart flushing.
+  test::ScopedFieldTrials field_trials(
+      "WebRTC-Audio-NetEqSmartFlushing/enabled:true,"
+      "target_level_threshold_ms:0,target_level_multiplier:2/");
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  // Insert 10 small packets; should be ok.
+  const int payload_len = 10;
+  int i;
+  for (i = 0; i < 10; ++i) {
+    EXPECT_EQ(
+        PacketBuffer::kOK,
+        buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                            /*stats=*/&mock_stats,
+                            /*last_decoded_length=*/payload_len,
+                            /*sample_rate=*/1000,
+                            /*target_level_ms=*/100,
+                            /*decoder_database=*/decoder_database));
+  }
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+  uint32_t next_ts;
+  EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&next_ts));
+  EXPECT_EQ(0u, next_ts);  // Expect first inserted packet to be first in line.
+
+  const Packet packet = gen.NextPacket(payload_len, nullptr);
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(6);
+  // Insert 11th packet; should cause a partial flush and insert the packet
+  // after flushing.
+  EXPECT_EQ(PacketBuffer::kPartialFlush,
+            buffer.InsertPacket(/*packet=*/packet.Clone(),
+                                /*stats=*/&mock_stats,
+                                /*last_decoded_length=*/payload_len,
+                                /*sample_rate=*/1000,
+                                /*target_level_ms=*/40,
+                                /*decoder_database=*/decoder_database));
+  EXPECT_EQ(5u, buffer.NumPacketsInBuffer());
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test inserting a list of packets.
+TEST(PacketBuffer, InsertPacketList) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  PacketList list;
+  const int payload_len = 10;
+
+  // Insert 10 small packets.
+  for (int i = 0; i < 10; ++i) {
+    list.push_back(gen.NextPacket(payload_len, nullptr));
+  }
+
+  MockDecoderDatabase decoder_database;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                          absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(0))
+      .WillRepeatedly(Return(&info));
+
+  StrictMock<MockStatisticsCalculator> mock_stats;
+
+  absl::optional<uint8_t> current_pt;
+  absl::optional<uint8_t> current_cng_pt;
+  EXPECT_EQ(
+      PacketBuffer::kOK,
+      buffer.InsertPacketList(/*packet_list=*/&list,
+                              /*decoder_database=*/decoder_database,
+                              /*current_rtp_payload_type=*/&current_pt,
+                              /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                              /*stats=*/&mock_stats,
+                              /*last_decoded_length=*/payload_len,
+                              /*sample_rate=*/1000,
+                              /*target_level_ms=*/30));
+  EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+  EXPECT_EQ(0, current_pt);  // Current payload type changed to 0.
+  EXPECT_EQ(absl::nullopt, current_cng_pt);  // CNG payload type not changed.
+
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test inserting a list of packets. Last packet is of a different payload type.
+// Expecting the buffer to flush.
+// TODO(hlundin): Remove this test when legacy operation is no longer needed.
+TEST(PacketBuffer, InsertPacketListChangePayloadType) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  PacketGenerator gen(0, 0, 0, 10);
+  PacketList list;
+  const int payload_len = 10;
+
+  // Insert 10 small packets.
+  for (int i = 0; i < 10; ++i) {
+    list.push_back(gen.NextPacket(payload_len, nullptr));
+  }
+  // Insert 11th packet of another payload type (not CNG).
+  {
+    Packet packet = gen.NextPacket(payload_len, nullptr);
+    packet.payload_type = 1;
+    list.push_back(std::move(packet));
+  }
+
+  MockDecoderDatabase decoder_database;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info0(SdpAudioFormat("pcmu", 8000, 1),
+                                           absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(0))
+      .WillRepeatedly(Return(&info0));
+  const DecoderDatabase::DecoderInfo info1(SdpAudioFormat("pcma", 8000, 1),
+                                           absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(1))
+      .WillRepeatedly(Return(&info1));
+
+  StrictMock<MockStatisticsCalculator> mock_stats;
+
+  absl::optional<uint8_t> current_pt;
+  absl::optional<uint8_t> current_cng_pt;
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1)).Times(10);
+  EXPECT_EQ(
+      PacketBuffer::kFlushed,
+      buffer.InsertPacketList(/*packet_list=*/&list,
+                              /*decoder_database=*/decoder_database,
+                              /*current_rtp_payload_type=*/&current_pt,
+                              /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                              /*stats=*/&mock_stats,
+                              /*last_decoded_length=*/payload_len,
+                              /*sample_rate=*/1000,
+                              /*target_level_ms=*/30));
+  EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());  // Only the last packet.
+  EXPECT_EQ(1, current_pt);  // Current payload type changed to 1.
+  EXPECT_EQ(absl::nullopt, current_cng_pt);  // CNG payload type not changed.
+
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+TEST(PacketBuffer, ExtractOrderRedundancy) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
+  const int kPackets = 18;
+  const int kFrameSize = 10;
+  const int kPayloadLength = 10;
+
+  PacketsToInsert packet_facts[kPackets] = {
+      {0xFFFD, 0xFFFFFFD7, 0, true, 0},   {0xFFFE, 0xFFFFFFE1, 0, true, 1},
+      {0xFFFE, 0xFFFFFFD7, 1, false, -1}, {0xFFFF, 0xFFFFFFEB, 0, true, 2},
+      {0xFFFF, 0xFFFFFFE1, 1, false, -1}, {0x0000, 0xFFFFFFF5, 0, true, 3},
+      {0x0000, 0xFFFFFFEB, 1, false, -1}, {0x0001, 0xFFFFFFFF, 0, true, 4},
+      {0x0001, 0xFFFFFFF5, 1, false, -1}, {0x0002, 0x0000000A, 0, true, 5},
+      {0x0002, 0xFFFFFFFF, 1, false, -1}, {0x0003, 0x0000000A, 1, false, -1},
+      {0x0004, 0x0000001E, 0, true, 7},   {0x0004, 0x00000014, 1, false, 6},
+      {0x0005, 0x0000001E, 0, true, -1},  {0x0005, 0x00000014, 1, false, -1},
+      {0x0006, 0x00000028, 0, true, 8},   {0x0006, 0x0000001E, 1, false, -1},
+  };
+  MockDecoderDatabase decoder_database;
+
+  const size_t kExpectPacketsInBuffer = 9;
+
+  std::vector<Packet> expect_order(kExpectPacketsInBuffer);
+
+  PacketGenerator gen(0, 0, 0, kFrameSize);
+
+  StrictMock<MockStatisticsCalculator> mock_stats;
+
+  // Interleaving the EXPECT_CALL sequence with expectations on the MockFunction
+  // check ensures that exactly one call to PacketsDiscarded happens in each
+  // DiscardNextPacket call.
+  InSequence s;
+  MockFunction<void(int check_point_id)> check;
+  for (int i = 0; i < kPackets; ++i) {
+    gen.Reset(packet_facts[i].sequence_number, packet_facts[i].timestamp,
+              packet_facts[i].payload_type, kFrameSize);
+    Packet packet = gen.NextPacket(kPayloadLength, nullptr);
+    packet.priority.codec_level = packet_facts[i].primary ? 0 : 1;
+    if (packet_facts[i].extract_order < 0) {
+      if (packet.priority.codec_level > 0) {
+        EXPECT_CALL(mock_stats, SecondaryPacketsDiscarded(1));
+      } else {
+        EXPECT_CALL(mock_stats, PacketsDiscarded(1));
+      }
+    }
+    EXPECT_CALL(check, Call(i));
+    EXPECT_EQ(PacketBuffer::kOK,
+              buffer.InsertPacket(/*packet=*/packet.Clone(),
+                                  /*stats=*/&mock_stats,
+                                  /*last_decoded_length=*/kPayloadLength,
+                                  /*sample_rate=*/1000,
+                                  /*target_level_ms=*/60,
+                                  /*decoder_database=*/decoder_database));
+    if (packet_facts[i].extract_order >= 0) {
+      expect_order[packet_facts[i].extract_order] = std::move(packet);
+    }
+    check.Call(i);
+  }
+
+  EXPECT_EQ(kExpectPacketsInBuffer, buffer.NumPacketsInBuffer());
+
+  for (size_t i = 0; i < kExpectPacketsInBuffer; ++i) {
+    const absl::optional<Packet> packet = buffer.GetNextPacket();
+    EXPECT_EQ(packet, expect_order[i]);  // Compare contents.
+  }
+  EXPECT_TRUE(buffer.Empty());
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+TEST(PacketBuffer, DiscardPackets) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
+  const uint16_t start_seq_no = 17;
+  const uint32_t start_ts = 4711;
+  const uint32_t ts_increment = 10;
+  PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment);
+  PacketList list;
+  const int payload_len = 10;
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  constexpr int kTotalPackets = 10;
+  // Insert 10 small packets.
+  for (int i = 0; i < kTotalPackets; ++i) {
+    buffer.InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                        /*stats=*/&mock_stats,
+                        /*last_decoded_length=*/payload_len,
+                        /*sample_rate=*/1000,
+                        /*target_level_ms=*/60,
+                        /*decoder_database=*/decoder_database);
+  }
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+
+  uint32_t current_ts = start_ts;
+
+  // Discard them one by one and make sure that the right packets are at the
+  // front of the buffer.
+  constexpr int kDiscardPackets = 5;
+
+  // Interleaving the EXPECT_CALL sequence with expectations on the MockFunction
+  // check ensures that exactly one call to PacketsDiscarded happens in each
+  // DiscardNextPacket call.
+  InSequence s;
+  MockFunction<void(int check_point_id)> check;
+  for (int i = 0; i < kDiscardPackets; ++i) {
+    uint32_t ts;
+    EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&ts));
+    EXPECT_EQ(current_ts, ts);
+    EXPECT_CALL(mock_stats, PacketsDiscarded(1));
+    EXPECT_CALL(check, Call(i));
+    EXPECT_EQ(PacketBuffer::kOK, buffer.DiscardNextPacket(&mock_stats));
+    current_ts += ts_increment;
+    check.Call(i);
+  }
+
+  constexpr int kRemainingPackets = kTotalPackets - kDiscardPackets;
+  // This will discard all remaining packets but one. The oldest packet is older
+  // than the indicated horizon_samples, and will thus be left in the buffer.
+  constexpr size_t kSkipPackets = 1;
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1))
+      .Times(kRemainingPackets - kSkipPackets);
+  EXPECT_CALL(check, Call(17));  // Arbitrary id number.
+  buffer.DiscardOldPackets(start_ts + kTotalPackets * ts_increment,
+                           kRemainingPackets * ts_increment, &mock_stats);
+  check.Call(17);  // Same arbitrary id number.
+
+  EXPECT_EQ(kSkipPackets, buffer.NumPacketsInBuffer());
+  uint32_t ts;
+  EXPECT_EQ(PacketBuffer::kOK, buffer.NextTimestamp(&ts));
+  EXPECT_EQ(current_ts, ts);
+
+  // Discard all remaining packets.
+  EXPECT_CALL(mock_stats, PacketsDiscarded(kSkipPackets));
+  buffer.DiscardAllOldPackets(start_ts + kTotalPackets * ts_increment,
+                              &mock_stats);
+
+  EXPECT_TRUE(buffer.Empty());
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+TEST(PacketBuffer, Reordering) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(100, &tick_timer);  // 100 packets.
+  const uint16_t start_seq_no = 17;
+  const uint32_t start_ts = 4711;
+  const uint32_t ts_increment = 10;
+  PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment);
+  const int payload_len = 10;
+
+  // Generate 10 small packets and insert them into a PacketList. Insert every
+  // odd packet to the front, and every even packet to the back, thus creating
+  // a (rather strange) reordering.
+  PacketList list;
+  for (int i = 0; i < 10; ++i) {
+    Packet packet = gen.NextPacket(payload_len, nullptr);
+    if (i % 2) {
+      list.push_front(std::move(packet));
+    } else {
+      list.push_back(std::move(packet));
+    }
+  }
+
+  MockDecoderDatabase decoder_database;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                          absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(0))
+      .WillRepeatedly(Return(&info));
+  absl::optional<uint8_t> current_pt;
+  absl::optional<uint8_t> current_cng_pt;
+
+  StrictMock<MockStatisticsCalculator> mock_stats;
+
+  EXPECT_EQ(
+      PacketBuffer::kOK,
+      buffer.InsertPacketList(/*packet_list=*/&list,
+                              /*decoder_database=*/decoder_database,
+                              /*current_rtp_payload_type=*/&current_pt,
+                              /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                              /*stats=*/&mock_stats,
+                              /*last_decoded_length=*/payload_len,
+                              /*sample_rate=*/1000,
+                              /*target_level_ms=*/30));
+  EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
+
+  // Extract them and make sure that come out in the right order.
+  uint32_t current_ts = start_ts;
+  for (int i = 0; i < 10; ++i) {
+    const absl::optional<Packet> packet = buffer.GetNextPacket();
+    ASSERT_TRUE(packet);
+    EXPECT_EQ(current_ts, packet->timestamp);
+    current_ts += ts_increment;
+  }
+  EXPECT_TRUE(buffer.Empty());
+
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// The test first inserts a packet with narrow-band CNG, then a packet with
+// wide-band speech. The expected behavior of the packet buffer is to detect a
+// change in sample rate, even though no speech packet has been inserted before,
+// and flush out the CNG packet.
+TEST(PacketBuffer, CngFirstThenSpeechWithNewSampleRate) {
+  TickTimer tick_timer;
+  PacketBuffer buffer(10, &tick_timer);  // 10 packets.
+  const uint8_t kCngPt = 13;
+  const int kPayloadLen = 10;
+  const uint8_t kSpeechPt = 100;
+
+  MockDecoderDatabase decoder_database;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info_cng(SdpAudioFormat("cn", 8000, 1),
+                                              absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(kCngPt))
+      .WillRepeatedly(Return(&info_cng));
+  const DecoderDatabase::DecoderInfo info_speech(
+      SdpAudioFormat("l16", 16000, 1), absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(kSpeechPt))
+      .WillRepeatedly(Return(&info_speech));
+
+  // Insert first packet, which is narrow-band CNG.
+  PacketGenerator gen(0, 0, kCngPt, 10);
+  PacketList list;
+  list.push_back(gen.NextPacket(kPayloadLen, nullptr));
+  absl::optional<uint8_t> current_pt;
+  absl::optional<uint8_t> current_cng_pt;
+
+  StrictMock<MockStatisticsCalculator> mock_stats;
+
+  EXPECT_EQ(
+      PacketBuffer::kOK,
+      buffer.InsertPacketList(/*packet_list=*/&list,
+                              /*decoder_database=*/decoder_database,
+                              /*current_rtp_payload_type=*/&current_pt,
+                              /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                              /*stats=*/&mock_stats,
+                              /*last_decoded_length=*/kPayloadLen,
+                              /*sample_rate=*/1000,
+                              /*target_level_ms=*/30));
+  EXPECT_TRUE(list.empty());
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
+  ASSERT_TRUE(buffer.PeekNextPacket());
+  EXPECT_EQ(kCngPt, buffer.PeekNextPacket()->payload_type);
+  EXPECT_EQ(current_pt, absl::nullopt);  // Current payload type not set.
+  EXPECT_EQ(kCngPt, current_cng_pt);     // CNG payload type set.
+
+  // Insert second packet, which is wide-band speech.
+  {
+    Packet packet = gen.NextPacket(kPayloadLen, nullptr);
+    packet.payload_type = kSpeechPt;
+    list.push_back(std::move(packet));
+  }
+  // Expect the buffer to flush out the CNG packet, since it does not match the
+  // new speech sample rate.
+  EXPECT_CALL(mock_stats, PacketsDiscarded(1));
+  EXPECT_EQ(
+      PacketBuffer::kFlushed,
+      buffer.InsertPacketList(/*packet_list=*/&list,
+                              /*decoder_database=*/decoder_database,
+                              /*current_rtp_payload_type=*/&current_pt,
+                              /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                              /*stats=*/&mock_stats,
+                              /*last_decoded_length=*/kPayloadLen,
+                              /*sample_rate=*/1000,
+                              /*target_level_ms=*/30));
+  EXPECT_TRUE(list.empty());
+  EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
+  ASSERT_TRUE(buffer.PeekNextPacket());
+  EXPECT_EQ(kSpeechPt, buffer.PeekNextPacket()->payload_type);
+
+  EXPECT_EQ(kSpeechPt, current_pt);          // Current payload type set.
+  EXPECT_EQ(absl::nullopt, current_cng_pt);  // CNG payload type reset.
+
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+TEST(PacketBuffer, Failures) {
+  const uint16_t start_seq_no = 17;
+  const uint32_t start_ts = 4711;
+  const uint32_t ts_increment = 10;
+  int payload_len = 100;
+  PacketGenerator gen(start_seq_no, start_ts, 0, ts_increment);
+  TickTimer tick_timer;
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  PacketBuffer* buffer = new PacketBuffer(100, &tick_timer);  // 100 packets.
+  {
+    Packet packet = gen.NextPacket(payload_len, nullptr);
+    packet.payload.Clear();
+    EXPECT_EQ(PacketBuffer::kInvalidPacket,
+              buffer->InsertPacket(/*packet=*/std::move(packet),
+                                   /*stats=*/&mock_stats,
+                                   /*last_decoded_length=*/payload_len,
+                                   /*sample_rate=*/1000,
+                                   /*target_level_ms=*/60,
+                                   /*decoder_database=*/decoder_database));
+  }
+  // Buffer should still be empty. Test all empty-checks.
+  uint32_t temp_ts;
+  EXPECT_EQ(PacketBuffer::kBufferEmpty, buffer->NextTimestamp(&temp_ts));
+  EXPECT_EQ(PacketBuffer::kBufferEmpty,
+            buffer->NextHigherTimestamp(0, &temp_ts));
+  EXPECT_EQ(NULL, buffer->PeekNextPacket());
+  EXPECT_FALSE(buffer->GetNextPacket());
+
+  // Discarding packets will not invoke mock_stats.PacketDiscarded() because the
+  // packet buffer is empty.
+  EXPECT_EQ(PacketBuffer::kBufferEmpty, buffer->DiscardNextPacket(&mock_stats));
+  buffer->DiscardAllOldPackets(0, &mock_stats);
+
+  // Insert one packet to make the buffer non-empty.
+  EXPECT_EQ(
+      PacketBuffer::kOK,
+      buffer->InsertPacket(/*packet=*/gen.NextPacket(payload_len, nullptr),
+                           /*stats=*/&mock_stats,
+                           /*last_decoded_length=*/payload_len,
+                           /*sample_rate=*/1000,
+                           /*target_level_ms=*/60,
+                           /*decoder_database=*/decoder_database));
+  EXPECT_EQ(PacketBuffer::kInvalidPointer, buffer->NextTimestamp(NULL));
+  EXPECT_EQ(PacketBuffer::kInvalidPointer,
+            buffer->NextHigherTimestamp(0, NULL));
+  delete buffer;
+
+  // Insert packet list of three packets, where the second packet has an invalid
+  // payload.  Expect first packet to be inserted, and the remaining two to be
+  // discarded.
+  buffer = new PacketBuffer(100, &tick_timer);  // 100 packets.
+  PacketList list;
+  list.push_back(gen.NextPacket(payload_len, nullptr));  // Valid packet.
+  {
+    Packet packet = gen.NextPacket(payload_len, nullptr);
+    packet.payload.Clear();  // Invalid.
+    list.push_back(std::move(packet));
+  }
+  list.push_back(gen.NextPacket(payload_len, nullptr));  // Valid packet.
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                          absl::nullopt, factory.get());
+  EXPECT_CALL(decoder_database, GetDecoderInfo(0))
+      .WillRepeatedly(Return(&info));
+  absl::optional<uint8_t> current_pt;
+  absl::optional<uint8_t> current_cng_pt;
+  EXPECT_EQ(
+      PacketBuffer::kInvalidPacket,
+      buffer->InsertPacketList(/*packet_list=*/&list,
+                               /*decoder_database=*/decoder_database,
+                               /*current_rtp_payload_type=*/&current_pt,
+                               /*current_cng_rtp_payload_type=*/&current_cng_pt,
+                               /*stats=*/&mock_stats,
+                               /*last_decoded_length=*/payload_len,
+                               /*sample_rate=*/1000,
+                               /*target_level_ms=*/30));
+  EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
+  EXPECT_EQ(1u, buffer->NumPacketsInBuffer());
+  delete buffer;
+  EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
+}
+
+// Test packet comparison function.
+// The function should return true if the first packet "goes before" the second.
+TEST(PacketBuffer, ComparePackets) {
+  PacketGenerator gen(0, 0, 0, 10);
+  Packet a(gen.NextPacket(10, nullptr));  // SN = 0, TS = 0.
+  Packet b(gen.NextPacket(10, nullptr));  // SN = 1, TS = 10.
+  EXPECT_FALSE(a == b);
+  EXPECT_TRUE(a != b);
+  EXPECT_TRUE(a < b);
+  EXPECT_FALSE(a > b);
+  EXPECT_TRUE(a <= b);
+  EXPECT_FALSE(a >= b);
+
+  // Testing wrap-around case; 'a' is earlier but has a larger timestamp value.
+  a.timestamp = 0xFFFFFFFF - 10;
+  EXPECT_FALSE(a == b);
+  EXPECT_TRUE(a != b);
+  EXPECT_TRUE(a < b);
+  EXPECT_FALSE(a > b);
+  EXPECT_TRUE(a <= b);
+  EXPECT_FALSE(a >= b);
+
+  // Test equal packets.
+  EXPECT_TRUE(a == a);
+  EXPECT_FALSE(a != a);
+  EXPECT_FALSE(a < a);
+  EXPECT_FALSE(a > a);
+  EXPECT_TRUE(a <= a);
+  EXPECT_TRUE(a >= a);
+
+  // Test equal timestamps but different sequence numbers (0 and 1).
+  a.timestamp = b.timestamp;
+  EXPECT_FALSE(a == b);
+  EXPECT_TRUE(a != b);
+  EXPECT_TRUE(a < b);
+  EXPECT_FALSE(a > b);
+  EXPECT_TRUE(a <= b);
+  EXPECT_FALSE(a >= b);
+
+  // Test equal timestamps but different sequence numbers (32767 and 1).
+  a.sequence_number = 0xFFFF;
+  EXPECT_FALSE(a == b);
+  EXPECT_TRUE(a != b);
+  EXPECT_TRUE(a < b);
+  EXPECT_FALSE(a > b);
+  EXPECT_TRUE(a <= b);
+  EXPECT_FALSE(a >= b);
+
+  // Test equal timestamps and sequence numbers, but differing priorities.
+  a.sequence_number = b.sequence_number;
+  a.priority = {1, 0};
+  b.priority = {0, 0};
+  // a after b
+  EXPECT_FALSE(a == b);
+  EXPECT_TRUE(a != b);
+  EXPECT_FALSE(a < b);
+  EXPECT_TRUE(a > b);
+  EXPECT_FALSE(a <= b);
+  EXPECT_TRUE(a >= b);
+
+  Packet c(gen.NextPacket(0, nullptr));  // SN = 2, TS = 20.
+  Packet d(gen.NextPacket(0, nullptr));  // SN = 3, TS = 20.
+  c.timestamp = b.timestamp;
+  d.timestamp = b.timestamp;
+  c.sequence_number = b.sequence_number;
+  d.sequence_number = b.sequence_number;
+  c.priority = {1, 1};
+  d.priority = {0, 1};
+  // c after d
+  EXPECT_FALSE(c == d);
+  EXPECT_TRUE(c != d);
+  EXPECT_FALSE(c < d);
+  EXPECT_TRUE(c > d);
+  EXPECT_FALSE(c <= d);
+  EXPECT_TRUE(c >= d);
+
+  // c after a
+  EXPECT_FALSE(c == a);
+  EXPECT_TRUE(c != a);
+  EXPECT_FALSE(c < a);
+  EXPECT_TRUE(c > a);
+  EXPECT_FALSE(c <= a);
+  EXPECT_TRUE(c >= a);
+
+  // c after b
+  EXPECT_FALSE(c == b);
+  EXPECT_TRUE(c != b);
+  EXPECT_FALSE(c < b);
+  EXPECT_TRUE(c > b);
+  EXPECT_FALSE(c <= b);
+  EXPECT_TRUE(c >= b);
+
+  // a after d
+  EXPECT_FALSE(a == d);
+  EXPECT_TRUE(a != d);
+  EXPECT_FALSE(a < d);
+  EXPECT_TRUE(a > d);
+  EXPECT_FALSE(a <= d);
+  EXPECT_TRUE(a >= d);
+
+  // d after b
+  EXPECT_FALSE(d == b);
+  EXPECT_TRUE(d != b);
+  EXPECT_FALSE(d < b);
+  EXPECT_TRUE(d > b);
+  EXPECT_FALSE(d <= b);
+  EXPECT_TRUE(d >= b);
+}
+
+TEST(PacketBuffer, GetSpanSamples) {
+  constexpr size_t kFrameSizeSamples = 10;
+  constexpr int kPayloadSizeBytes = 1;  // Does not matter to this test;
+  constexpr uint32_t kStartTimeStamp = 0xFFFFFFFE;  // Close to wrap around.
+  constexpr int kSampleRateHz = 48000;
+  constexpr bool KCountDtxWaitingTime = false;
+  TickTimer tick_timer;
+  PacketBuffer buffer(3, &tick_timer);
+  PacketGenerator gen(0, kStartTimeStamp, 0, kFrameSizeSamples);
+  StrictMock<MockStatisticsCalculator> mock_stats;
+  MockDecoderDatabase decoder_database;
+
+  Packet packet_1 = gen.NextPacket(kPayloadSizeBytes, nullptr);
+
+  std::unique_ptr<MockEncodedAudioFrame> mock_audio_frame =
+      std::make_unique<MockEncodedAudioFrame>();
+  EXPECT_CALL(*mock_audio_frame, Duration())
+      .WillRepeatedly(Return(kFrameSizeSamples));
+  Packet packet_2 =
+      gen.NextPacket(kPayloadSizeBytes, std::move(mock_audio_frame));
+
+  RTC_DCHECK_GT(packet_1.timestamp,
+                packet_2.timestamp);  // Tmestamp wrapped around.
+
+  EXPECT_EQ(PacketBuffer::kOK,
+            buffer.InsertPacket(/*packet=*/std::move(packet_1),
+                                /*stats=*/&mock_stats,
+                                /*last_decoded_length=*/kFrameSizeSamples,
+                                /*sample_rate=*/1000,
+                                /*target_level_ms=*/60,
+                                /*decoder_database=*/decoder_database));
+
+  constexpr size_t kLastDecodedSizeSamples = 2;
+  // packet_1 has no access to duration, and relies last decoded duration as
+  // input.
+  EXPECT_EQ(kLastDecodedSizeSamples,
+            buffer.GetSpanSamples(kLastDecodedSizeSamples, kSampleRateHz,
+                                  KCountDtxWaitingTime));
+
+  EXPECT_EQ(PacketBuffer::kOK,
+            buffer.InsertPacket(/*packet=*/std::move(packet_2),
+                                /*stats=*/&mock_stats,
+                                /*last_decoded_length=*/kFrameSizeSamples,
+                                /*sample_rate=*/1000,
+                                /*target_level_ms=*/60,
+                                /*decoder_database=*/decoder_database));
+
+  EXPECT_EQ(kFrameSizeSamples * 2,
+            buffer.GetSpanSamples(0, kSampleRateHz, KCountDtxWaitingTime));
+
+  // packet_2 has access to duration, and ignores last decoded duration as
+  // input.
+  EXPECT_EQ(kFrameSizeSamples * 2,
+            buffer.GetSpanSamples(kLastDecodedSizeSamples, kSampleRateHz,
+                                  KCountDtxWaitingTime));
+}
+
+namespace {
+void TestIsObsoleteTimestamp(uint32_t limit_timestamp) {
+  // Check with zero horizon, which implies that the horizon is at 2^31, i.e.,
+  // half the timestamp range.
+  static const uint32_t kZeroHorizon = 0;
+  static const uint32_t k2Pow31Minus1 = 0x7FFFFFFF;
+  // Timestamp on the limit is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
+      limit_timestamp, limit_timestamp, kZeroHorizon));
+  // 1 sample behind is old.
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 1,
+                                                limit_timestamp, kZeroHorizon));
+  // 2^31 - 1 samples behind is old.
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - k2Pow31Minus1,
+                                                limit_timestamp, kZeroHorizon));
+  // 1 sample ahead is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
+      limit_timestamp + 1, limit_timestamp, kZeroHorizon));
+  // If |t1-t2|=2^31 and t1>t2, t2 is older than t1 but not the opposite.
+  uint32_t other_timestamp = limit_timestamp + (1 << 31);
+  uint32_t lowest_timestamp = std::min(limit_timestamp, other_timestamp);
+  uint32_t highest_timestamp = std::max(limit_timestamp, other_timestamp);
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(
+      lowest_timestamp, highest_timestamp, kZeroHorizon));
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
+      highest_timestamp, lowest_timestamp, kZeroHorizon));
+
+  // Fixed horizon at 10 samples.
+  static const uint32_t kHorizon = 10;
+  // Timestamp on the limit is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp,
+                                                 limit_timestamp, kHorizon));
+  // 1 sample behind is old.
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 1,
+                                                limit_timestamp, kHorizon));
+  // 9 samples behind is old.
+  EXPECT_TRUE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 9,
+                                                limit_timestamp, kHorizon));
+  // 10 samples behind is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp - 10,
+                                                 limit_timestamp, kHorizon));
+  // 2^31 - 1 samples behind is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(
+      limit_timestamp - k2Pow31Minus1, limit_timestamp, kHorizon));
+  // 1 sample ahead is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp + 1,
+                                                 limit_timestamp, kHorizon));
+  // 2^31 samples ahead is not old.
+  EXPECT_FALSE(PacketBuffer::IsObsoleteTimestamp(limit_timestamp + (1 << 31),
+                                                 limit_timestamp, kHorizon));
+}
+}  // namespace
+
+// Test the IsObsoleteTimestamp method with different limit timestamps.
+TEST(PacketBuffer, IsObsoleteTimestamp) {
+  TestIsObsoleteTimestamp(0);
+  TestIsObsoleteTimestamp(1);
+  TestIsObsoleteTimestamp(0xFFFFFFFF);  // -1 in uint32_t.
+  TestIsObsoleteTimestamp(0x80000000);  // 2^31.
+  TestIsObsoleteTimestamp(0x80000001);  // 2^31 + 1.
+  TestIsObsoleteTimestamp(0x7FFFFFFF);  // 2^31 - 1.
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc
new file mode 100644
index 0000000000..9999d6764b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc
@@ -0,0 +1,90 @@
+/*
+ *  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 "modules/audio_coding/neteq/post_decode_vad.h"
+
+namespace webrtc {
+
+PostDecodeVad::~PostDecodeVad() {
+  if (vad_instance_)
+    WebRtcVad_Free(vad_instance_);
+}
+
+void PostDecodeVad::Enable() {
+  if (!vad_instance_) {
+    // Create the instance.
+    vad_instance_ = WebRtcVad_Create();
+    if (vad_instance_ == nullptr) {
+      // Failed to create instance.
+      Disable();
+      return;
+    }
+  }
+  Init();
+  enabled_ = true;
+}
+
+void PostDecodeVad::Disable() {
+  enabled_ = false;
+  running_ = false;
+}
+
+void PostDecodeVad::Init() {
+  running_ = false;
+  if (vad_instance_) {
+    WebRtcVad_Init(vad_instance_);
+    WebRtcVad_set_mode(vad_instance_, kVadMode);
+    running_ = true;
+  }
+}
+
+void PostDecodeVad::Update(int16_t* signal,
+                           size_t length,
+                           AudioDecoder::SpeechType speech_type,
+                           bool sid_frame,
+                           int fs_hz) {
+  if (!vad_instance_ || !enabled_) {
+    return;
+  }
+
+  if (speech_type == AudioDecoder::kComfortNoise || sid_frame ||
+      fs_hz > 16000) {
+    // TODO(hlundin): Remove restriction on fs_hz.
+    running_ = false;
+    active_speech_ = true;
+    sid_interval_counter_ = 0;
+  } else if (!running_) {
+    ++sid_interval_counter_;
+  }
+
+  if (sid_interval_counter_ >= kVadAutoEnable) {
+    Init();
+  }
+
+  if (length > 0 && running_) {
+    size_t vad_sample_index = 0;
+    active_speech_ = false;
+    // Loop through frame sizes 30, 20, and 10 ms.
+    for (int vad_frame_size_ms = 30; vad_frame_size_ms >= 10;
+         vad_frame_size_ms -= 10) {
+      size_t vad_frame_size_samples =
+          static_cast<size_t>(vad_frame_size_ms * fs_hz / 1000);
+      while (length - vad_sample_index >= vad_frame_size_samples) {
+        int vad_return =
+            WebRtcVad_Process(vad_instance_, fs_hz, &signal[vad_sample_index],
+                              vad_frame_size_samples);
+        active_speech_ |= (vad_return == 1);
+        vad_sample_index += vad_frame_size_samples;
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.h b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.h
new file mode 100644
index 0000000000..3bd91b9edb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.h
@@ -0,0 +1,71 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
+#define MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "common_audio/vad/include/webrtc_vad.h"
+
+namespace webrtc {
+
+class PostDecodeVad {
+ public:
+  PostDecodeVad()
+      : enabled_(false),
+        running_(false),
+        active_speech_(true),
+        sid_interval_counter_(0),
+        vad_instance_(NULL) {}
+
+  virtual ~PostDecodeVad();
+
+  PostDecodeVad(const PostDecodeVad&) = delete;
+  PostDecodeVad& operator=(const PostDecodeVad&) = delete;
+
+  // Enables post-decode VAD.
+  void Enable();
+
+  // Disables post-decode VAD.
+  void Disable();
+
+  // Initializes post-decode VAD.
+  void Init();
+
+  // Updates post-decode VAD with the audio data in `signal` having `length`
+  // samples. The data is of type `speech_type`, at the sample rate `fs_hz`.
+  void Update(int16_t* signal,
+              size_t length,
+              AudioDecoder::SpeechType speech_type,
+              bool sid_frame,
+              int fs_hz);
+
+  // Accessors.
+  bool enabled() const { return enabled_; }
+  bool running() const { return running_; }
+  bool active_speech() const { return active_speech_; }
+
+ private:
+  static const int kVadMode = 0;  // Sets aggressiveness to "Normal".
+  // Number of Update() calls without CNG/SID before re-enabling VAD.
+  static const int kVadAutoEnable = 3000;
+
+  bool enabled_;
+  bool running_;
+  bool active_speech_;
+  int sid_interval_counter_;
+  ::VadInst* vad_instance_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc
new file mode 100644
index 0000000000..da3e4e864e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc
@@ -0,0 +1,25 @@
+/*
+ *  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.
+ */
+
+// Unit tests for PostDecodeVad class.
+
+#include "modules/audio_coding/neteq/post_decode_vad.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(PostDecodeVad, CreateAndDestroy) {
+  PostDecodeVad vad;
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.cc b/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.cc
new file mode 100644
index 0000000000..232170b177
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.cc
@@ -0,0 +1,117 @@
+/*
+ *  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/neteq/preemptive_expand.h"
+
+#include <algorithm>
+
+#include "api/array_view.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/time_stretch.h"
+
+namespace webrtc {
+
+PreemptiveExpand::ReturnCodes PreemptiveExpand::Process(
+    const int16_t* input,
+    size_t input_length,
+    size_t old_data_length,
+    AudioMultiVector* output,
+    size_t* length_change_samples) {
+  old_data_length_per_channel_ = old_data_length;
+  // Input length must be (almost) 30 ms.
+  // Also, the new part must be at least `overlap_samples_` elements.
+  static const size_t k15ms = 120;  // 15 ms = 120 samples at 8 kHz sample rate.
+  if (num_channels_ == 0 ||
+      input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_ ||
+      old_data_length >= input_length / num_channels_ - overlap_samples_) {
+    // Length of input data too short to do preemptive expand. Simply move all
+    // data from input to output.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, input_length));
+    return kError;
+  }
+  const bool kFastMode = false;  // Fast mode is not available for PE Expand.
+  return TimeStretch::Process(input, input_length, kFastMode, output,
+                              length_change_samples);
+}
+
+void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len,
+                                                     int16_t* best_correlation,
+                                                     size_t* peak_index) const {
+  // When the signal does not contain any active speech, the correlation does
+  // not matter. Simply set it to zero.
+  *best_correlation = 0;
+
+  // For low energy expansion, the new data can be less than 15 ms,
+  // but we must ensure that best_correlation is not larger than the length of
+  // the new data.
+  // but we must ensure that best_correlation is not larger than the new data.
+  *peak_index = std::min(*peak_index, len - old_data_length_per_channel_);
+}
+
+PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
+    const int16_t* input,
+    size_t input_length,
+    size_t peak_index,
+    int16_t best_correlation,
+    bool active_speech,
+    bool /*fast_mode*/,
+    AudioMultiVector* output) const {
+  // Pre-calculate common multiplication with `fs_mult_`.
+  // 120 corresponds to 15 ms.
+  size_t fs_mult_120 = static_cast<size_t>(fs_mult_ * 120);
+  // Check for strong correlation (>0.9 in Q14) and at least 15 ms new data,
+  // or passive speech.
+  if (((best_correlation > kCorrelationThreshold) &&
+       (old_data_length_per_channel_ <= fs_mult_120)) ||
+      !active_speech) {
+    // Do accelerate operation by overlap add.
+
+    // Set length of the first part, not to be modified.
+    size_t unmodified_length =
+        std::max(old_data_length_per_channel_, fs_mult_120);
+    // Copy first part, including cross-fade region.
+    output->PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        input, (unmodified_length + peak_index) * num_channels_));
+    // Copy the last `peak_index` samples up to 15 ms to `temp_vector`.
+    AudioMultiVector temp_vector(num_channels_);
+    temp_vector.PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        &input[(unmodified_length - peak_index) * num_channels_],
+        peak_index * num_channels_));
+    // Cross-fade `temp_vector` onto the end of `output`.
+    output->CrossFade(temp_vector, peak_index);
+    // Copy the last unmodified part, 15 ms + pitch period until the end.
+    output->PushBackInterleaved(rtc::ArrayView<const int16_t>(
+        &input[unmodified_length * num_channels_],
+        input_length - unmodified_length * num_channels_));
+
+    if (active_speech) {
+      return kSuccess;
+    } else {
+      return kSuccessLowEnergy;
+    }
+  } else {
+    // Accelerate not allowed. Simply move all data from decoded to outData.
+    output->PushBackInterleaved(
+        rtc::ArrayView<const int16_t>(input, input_length));
+    return kNoStretch;
+  }
+}
+
+PreemptiveExpand* PreemptiveExpandFactory::Create(
+    int sample_rate_hz,
+    size_t num_channels,
+    const BackgroundNoise& background_noise,
+    size_t overlap_samples) const {
+  return new PreemptiveExpand(sample_rate_hz, num_channels, background_noise,
+                              overlap_samples);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.h b/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.h
new file mode 100644
index 0000000000..6338b993fd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.h
@@ -0,0 +1,85 @@
+/*
+ *  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_NETEQ_PREEMPTIVE_EXPAND_H_
+#define MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/neteq/time_stretch.h"
+
+namespace webrtc {
+
+class AudioMultiVector;
+class BackgroundNoise;
+
+// This class implements the PreemptiveExpand operation. Most of the work is
+// done in the base class TimeStretch, which is shared with the Accelerate
+// operation. In the PreemptiveExpand class, the operations that are specific to
+// PreemptiveExpand are implemented.
+class PreemptiveExpand : public TimeStretch {
+ public:
+  PreemptiveExpand(int sample_rate_hz,
+                   size_t num_channels,
+                   const BackgroundNoise& background_noise,
+                   size_t overlap_samples)
+      : TimeStretch(sample_rate_hz, num_channels, background_noise),
+        old_data_length_per_channel_(0),
+        overlap_samples_(overlap_samples) {}
+
+  PreemptiveExpand(const PreemptiveExpand&) = delete;
+  PreemptiveExpand& operator=(const PreemptiveExpand&) = delete;
+
+  // This method performs the actual PreemptiveExpand operation. The samples are
+  // read from `input`, of length `input_length` elements, and are written to
+  // `output`. The number of samples added through time-stretching is
+  // is provided in the output `length_change_samples`. The method returns
+  // the outcome of the operation as an enumerator value.
+  ReturnCodes Process(const int16_t* pw16_decoded,
+                      size_t len,
+                      size_t old_data_len,
+                      AudioMultiVector* output,
+                      size_t* length_change_samples);
+
+ protected:
+  // Sets the parameters `best_correlation` and `peak_index` to suitable
+  // values when the signal contains no active speech.
+  void SetParametersForPassiveSpeech(size_t input_length,
+                                     int16_t* best_correlation,
+                                     size_t* peak_index) const override;
+
+  // Checks the criteria for performing the time-stretching operation and,
+  // if possible, performs the time-stretching.
+  ReturnCodes CheckCriteriaAndStretch(const int16_t* input,
+                                      size_t input_length,
+                                      size_t peak_index,
+                                      int16_t best_correlation,
+                                      bool active_speech,
+                                      bool /*fast_mode*/,
+                                      AudioMultiVector* output) const override;
+
+ private:
+  size_t old_data_length_per_channel_;
+  size_t overlap_samples_;
+};
+
+struct PreemptiveExpandFactory {
+  PreemptiveExpandFactory() {}
+  virtual ~PreemptiveExpandFactory() {}
+
+  virtual PreemptiveExpand* Create(int sample_rate_hz,
+                                   size_t num_channels,
+                                   const BackgroundNoise& background_noise,
+                                   size_t overlap_samples) const;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.cc b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.cc
new file mode 100644
index 0000000000..ada175831c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.cc
@@ -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.
+ */
+
+#include "modules/audio_coding/neteq/random_vector.h"
+
+namespace webrtc {
+
+const int16_t RandomVector::kRandomTable[RandomVector::kRandomTableSize] = {
+    2680,   5532,   441,    5520,   16170,  -5146,  -1024,  -8733,  3115,
+    9598,   -10380, -4959,  -1280,  -21716, 7133,   -1522,  13458,  -3902,
+    2789,   -675,   3441,   5016,   -13599, -4003,  -2739,  3922,   -7209,
+    13352,  -11617, -7241,  12905,  -2314,  5426,   10121,  -9702,  11207,
+    -13542, 1373,   816,    -5934,  -12504, 4798,   1811,   4112,   -613,
+    201,    -10367, -2960,  -2419,  3442,   4299,   -6116,  -6092,  1552,
+    -1650,  -480,   -1237,  18720,  -11858, -8303,  -8212,  865,    -2890,
+    -16968, 12052,  -5845,  -5912,  9777,   -5665,  -6294,  5426,   -4737,
+    -6335,  1652,   761,    3832,   641,    -8552,  -9084,  -5753,  8146,
+    12156,  -4915,  15086,  -1231,  -1869,  11749,  -9319,  -6403,  11407,
+    6232,   -1683,  24340,  -11166, 4017,   -10448, 3153,   -2936,  6212,
+    2891,   -866,   -404,   -4807,  -2324,  -1917,  -2388,  -6470,  -3895,
+    -10300, 5323,   -5403,  2205,   4640,   7022,   -21186, -6244,  -882,
+    -10031, -3395,  -12885, 7155,   -5339,  5079,   -2645,  -9515,  6622,
+    14651,  15852,  359,    122,    8246,   -3502,  -6696,  -3679,  -13535,
+    -1409,  -704,   -7403,  -4007,  1798,   279,    -420,   -12796, -14219,
+    1141,   3359,   11434,  7049,   -6684,  -7473,  14283,  -4115,  -9123,
+    -8969,  4152,   4117,   13792,  5742,   16168,  8661,   -1609,  -6095,
+    1881,   14380,  -5588,  6758,   -6425,  -22969, -7269,  7031,   1119,
+    -1611,  -5850,  -11281, 3559,   -8952,  -10146, -4667,  -16251, -1538,
+    2062,   -1012,  -13073, 227,    -3142,  -5265,  20,     5770,   -7559,
+    4740,   -4819,  992,    -8208,  -7130,  -4652,  6725,   7369,   -1036,
+    13144,  -1588,  -5304,  -2344,  -449,   -5705,  -8894,  5205,   -17904,
+    -11188, -1022,  4852,   10101,  -5255,  -4200,  -752,   7941,   -1543,
+    5959,   14719,  13346,  17045,  -15605, -1678,  -1600,  -9230,  68,
+    23348,  1172,   7750,   11212,  -18227, 9956,   4161,   883,    3947,
+    4341,   1014,   -4889,  -2603,  1246,   -5630,  -3596,  -870,   -1298,
+    2784,   -3317,  -6612,  -20541, 4166,   4181,   -8625,  3562,   12890,
+    4761,   3205,   -12259, -8579};
+
+void RandomVector::Reset() {
+  seed_ = 777;
+  seed_increment_ = 1;
+}
+
+void RandomVector::Generate(size_t length, int16_t* output) {
+  for (size_t i = 0; i < length; i++) {
+    seed_ += seed_increment_;
+    size_t position = seed_ & (kRandomTableSize - 1);
+    output[i] = kRandomTable[position];
+  }
+}
+
+void RandomVector::IncreaseSeedIncrement(int16_t increase_by) {
+  seed_increment_ += increase_by;
+  seed_increment_ &= kRandomTableSize - 1;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.h b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.h
new file mode 100644
index 0000000000..4a782f1116
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.h
@@ -0,0 +1,46 @@
+/*
+ *  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_NETEQ_RANDOM_VECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+// This class generates pseudo-random samples.
+class RandomVector {
+ public:
+  static const size_t kRandomTableSize = 256;
+  static const int16_t kRandomTable[kRandomTableSize];
+
+  RandomVector() : seed_(777), seed_increment_(1) {}
+
+  RandomVector(const RandomVector&) = delete;
+  RandomVector& operator=(const RandomVector&) = delete;
+
+  void Reset();
+
+  void Generate(size_t length, int16_t* output);
+
+  void IncreaseSeedIncrement(int16_t increase_by);
+
+  // Accessors and mutators.
+  int16_t seed_increment() { return seed_increment_; }
+  void set_seed_increment(int16_t value) { seed_increment_ = value; }
+
+ private:
+  uint32_t seed_;
+  int16_t seed_increment_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/random_vector_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector_unittest.cc
new file mode 100644
index 0000000000..44479a6dd6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/random_vector_unittest.cc
@@ -0,0 +1,25 @@
+/*
+ *  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.
+ */
+
+// Unit tests for RandomVector class.
+
+#include "modules/audio_coding/neteq/random_vector.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(RandomVector, CreateAndDestroy) {
+  RandomVector random_vector;
+}
+
+// TODO(hlundin): Write more tests.
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.cc
new file mode 100644
index 0000000000..cec9f2f8a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.cc
@@ -0,0 +1,190 @@
+/*
+ *  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/neteq/red_payload_splitter.h"
+
+#include <stddef.h>
+
+#include <cstdint>
+#include <list>
+#include <utility>
+#include <vector>
+
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+// The method loops through a list of packets {A, B, C, ...}. Each packet is
+// split into its corresponding RED payloads, {A1, A2, ...}, which is
+// temporarily held in the list `new_packets`.
+// When the first packet in `packet_list` has been processed, the original
+// packet is replaced by the new ones in `new_packets`, so that `packet_list`
+// becomes: {A1, A2, ..., B, C, ...}. The method then continues with B, and C,
+// until all the original packets have been replaced by their split payloads.
+bool RedPayloadSplitter::SplitRed(PacketList* packet_list) {
+  // Too many RED blocks indicates that something is wrong. Clamp it at some
+  // reasonable value.
+  const size_t kMaxRedBlocks = 32;
+  bool ret = true;
+  PacketList::iterator it = packet_list->begin();
+  while (it != packet_list->end()) {
+    const Packet& red_packet = *it;
+    RTC_DCHECK(!red_packet.payload.empty());
+    const uint8_t* payload_ptr = red_packet.payload.data();
+    size_t payload_length = red_packet.payload.size();
+
+    // Read RED headers (according to RFC 2198):
+    //
+    //    0                   1                   2                   3
+    //    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+    //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    //   |F|   block PT  |  timestamp offset         |   block length    |
+    //   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+    // Last RED header:
+    //    0 1 2 3 4 5 6 7
+    //   +-+-+-+-+-+-+-+-+
+    //   |0|   Block PT  |
+    //   +-+-+-+-+-+-+-+-+
+
+    struct RedHeader {
+      uint8_t payload_type;
+      uint32_t timestamp;
+      size_t payload_length;
+    };
+
+    std::vector<RedHeader> new_headers;
+    bool last_block = false;
+    size_t sum_length = 0;
+    while (!last_block) {
+      if (payload_length == 0) {
+        RTC_LOG(LS_WARNING) << "SplitRed header too short";
+        return false;
+      }
+      RedHeader new_header;
+      // Check the F bit. If F == 0, this was the last block.
+      last_block = ((*payload_ptr & 0x80) == 0);
+      // Bits 1 through 7 are payload type.
+      new_header.payload_type = payload_ptr[0] & 0x7F;
+      if (last_block) {
+        // No more header data to read.
+        sum_length += kRedLastHeaderLength;  // Account for RED header size.
+        new_header.timestamp = red_packet.timestamp;
+        new_header.payload_length = red_packet.payload.size() - sum_length;
+        payload_ptr += kRedLastHeaderLength;  // Advance to first payload byte.
+        payload_length -= kRedLastHeaderLength;
+      } else {
+        if (payload_length < kRedHeaderLength) {
+          RTC_LOG(LS_WARNING) << "SplitRed header too short";
+          return false;
+        }
+        // Bits 8 through 21 are timestamp offset.
+        int timestamp_offset =
+            (payload_ptr[1] << 6) + ((payload_ptr[2] & 0xFC) >> 2);
+        new_header.timestamp = red_packet.timestamp - timestamp_offset;
+        // Bits 22 through 31 are payload length.
+        new_header.payload_length =
+            ((payload_ptr[2] & 0x03) << 8) + payload_ptr[3];
+
+        sum_length += new_header.payload_length;
+        sum_length += kRedHeaderLength;  // Account for RED header size.
+
+        payload_ptr += kRedHeaderLength;  // Advance to next RED header.
+        payload_length -= kRedHeaderLength;
+      }
+      // Store in new list of packets.
+      if (new_header.payload_length > 0) {
+        new_headers.push_back(new_header);
+      }
+    }
+
+    if (new_headers.size() <= kMaxRedBlocks) {
+      // Populate the new packets with payload data.
+      // `payload_ptr` now points at the first payload byte.
+      PacketList new_packets;  // An empty list to store the split packets in.
+      for (size_t i = 0; i != new_headers.size(); ++i) {
+        const auto& new_header = new_headers[i];
+        size_t payload_length = new_header.payload_length;
+        if (payload_ptr + payload_length >
+            red_packet.payload.data() + red_packet.payload.size()) {
+          // The block lengths in the RED headers do not match the overall
+          // packet length. Something is corrupt. Discard this and the remaining
+          // payloads from this packet.
+          RTC_LOG(LS_WARNING) << "SplitRed length mismatch";
+          ret = false;
+          break;
+        }
+
+        Packet new_packet;
+        new_packet.timestamp = new_header.timestamp;
+        new_packet.payload_type = new_header.payload_type;
+        new_packet.sequence_number = red_packet.sequence_number;
+        new_packet.priority.red_level =
+            rtc::dchecked_cast<int>((new_headers.size() - 1) - i);
+        new_packet.payload.SetData(payload_ptr, payload_length);
+        new_packet.packet_info = RtpPacketInfo(
+            /*ssrc=*/red_packet.packet_info.ssrc(),
+            /*csrcs=*/std::vector<uint32_t>(),
+            /*rtp_timestamp=*/new_packet.timestamp,
+            /*receive_time=*/red_packet.packet_info.receive_time());
+        new_packet.packet_info.set_audio_level(
+            red_packet.packet_info.audio_level());
+        new_packets.push_front(std::move(new_packet));
+        payload_ptr += payload_length;
+      }
+      // Insert new packets into original list, before the element pointed to by
+      // iterator `it`.
+      packet_list->splice(it, std::move(new_packets));
+    } else {
+      RTC_LOG(LS_WARNING) << "SplitRed too many blocks: " << new_headers.size();
+      ret = false;
+    }
+    // Remove `it` from the packet list. This operation effectively moves the
+    // iterator `it` to the next packet in the list. Thus, we do not have to
+    // increment it manually.
+    it = packet_list->erase(it);
+  }
+  return ret;
+}
+
+void RedPayloadSplitter::CheckRedPayloads(
+    PacketList* packet_list,
+    const DecoderDatabase& decoder_database) {
+  int main_payload_type = -1;
+  for (auto it = packet_list->begin(); it != packet_list->end(); /* */) {
+    uint8_t this_payload_type = it->payload_type;
+    if (decoder_database.IsRed(this_payload_type)) {
+      it = packet_list->erase(it);
+      continue;
+    }
+    if (!decoder_database.IsDtmf(this_payload_type) &&
+        !decoder_database.IsComfortNoise(this_payload_type)) {
+      if (main_payload_type == -1) {
+        // This is the first packet in the list which is non-DTMF non-CNG.
+        main_payload_type = this_payload_type;
+      } else {
+        if (this_payload_type != main_payload_type) {
+          // We do not allow redundant payloads of a different type.
+          // Remove `it` from the packet list. This operation effectively
+          // moves the iterator `it` to the next packet in the list. Thus, we
+          // do not have to increment it manually.
+          it = packet_list->erase(it);
+          continue;
+        }
+      }
+    }
+    ++it;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.h b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.h
new file mode 100644
index 0000000000..2f48e4b7d4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.h
@@ -0,0 +1,51 @@
+/*
+ *  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_NETEQ_RED_PAYLOAD_SPLITTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
+
+#include "modules/audio_coding/neteq/packet.h"
+
+namespace webrtc {
+
+class DecoderDatabase;
+
+static const size_t kRedHeaderLength = 4;  // 4 bytes RED header.
+static const size_t kRedLastHeaderLength =
+    1;  // reduced size for last RED header.
+// This class handles splitting of RED payloads into smaller parts.
+// Codec-specific packet splitting can be performed by
+// AudioDecoder::ParsePayload.
+class RedPayloadSplitter {
+ public:
+  RedPayloadSplitter() {}
+
+  virtual ~RedPayloadSplitter() {}
+
+  RedPayloadSplitter(const RedPayloadSplitter&) = delete;
+  RedPayloadSplitter& operator=(const RedPayloadSplitter&) = delete;
+
+  // Splits each packet in `packet_list` into its separate RED payloads. Each
+  // RED payload is packetized into a Packet. The original elements in
+  // `packet_list` are properly deleted, and replaced by the new packets.
+  // Note that all packets in `packet_list` must be RED payloads, i.e., have
+  // RED headers according to RFC 2198 at the very beginning of the payload.
+  // Returns kOK or an error.
+  virtual bool SplitRed(PacketList* packet_list);
+
+  // Checks all packets in `packet_list`. Packets that are DTMF events or
+  // comfort noise payloads are kept. Except that, only one single payload type
+  // is accepted. Any packet with another payload type is discarded.
+  virtual void CheckRedPayloads(PacketList* packet_list,
+                                const DecoderDatabase& decoder_database);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
new file mode 100644
index 0000000000..a0ba5414ea
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
@@ -0,0 +1,390 @@
+/*
+ *  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.
+ */
+
+// Unit tests for RedPayloadSplitter class.
+
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
+
+
+#include <memory>
+#include <utility>  // pair
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder_factory.h"
+
+using ::testing::Return;
+using ::testing::ReturnNull;
+
+namespace webrtc {
+
+static const int kRedPayloadType = 100;
+static const size_t kPayloadLength = 10;
+static const uint16_t kSequenceNumber = 0;
+static const uint32_t kBaseTimestamp = 0x12345678;
+
+// A possible Opus packet that contains FEC is the following.
+// The frame is 20 ms in duration.
+//
+// 0                   1                   2                   3
+// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |0|0|0|0|1|0|0|0|x|1|x|x|x|x|x|x|x|                             |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                             |
+// |                    Compressed frame 1 (N-2 bytes)...          :
+// :                                                               |
+// |                                                               |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+void CreateOpusFecPayload(uint8_t* payload,
+                          size_t payload_length,
+                          uint8_t payload_value) {
+  if (payload_length < 2) {
+    return;
+  }
+  payload[0] = 0x08;
+  payload[1] = 0x40;
+  memset(&payload[2], payload_value, payload_length - 2);
+}
+
+// RED headers (according to RFC 2198):
+//
+//    0                   1                   2                   3
+//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//   |F|   block PT  |  timestamp offset         |   block length    |
+//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Last RED header:
+//    0 1 2 3 4 5 6 7
+//   +-+-+-+-+-+-+-+-+
+//   |0|   Block PT  |
+//   +-+-+-+-+-+-+-+-+
+
+// Creates a RED packet, with `num_payloads` payloads, with payload types given
+// by the values in array `payload_types` (which must be of length
+// `num_payloads`). Each redundant payload is `timestamp_offset` samples
+// "behind" the the previous payload.
+Packet CreateRedPayload(size_t num_payloads,
+                        uint8_t* payload_types,
+                        int timestamp_offset,
+                        bool embed_opus_fec = false) {
+  Packet packet;
+  packet.payload_type = kRedPayloadType;
+  packet.timestamp = kBaseTimestamp;
+  packet.sequence_number = kSequenceNumber;
+  packet.payload.SetSize((kPayloadLength + 1) +
+                         (num_payloads - 1) *
+                             (kPayloadLength + kRedHeaderLength));
+  uint8_t* payload_ptr = packet.payload.data();
+  for (size_t i = 0; i < num_payloads; ++i) {
+    // Write the RED headers.
+    if (i == num_payloads - 1) {
+      // Special case for last payload.
+      *payload_ptr = payload_types[i] & 0x7F;  // F = 0;
+      ++payload_ptr;
+      break;
+    }
+    *payload_ptr = payload_types[i] & 0x7F;
+    // Not the last block; set F = 1.
+    *payload_ptr |= 0x80;
+    ++payload_ptr;
+    int this_offset =
+        rtc::checked_cast<int>((num_payloads - i - 1) * timestamp_offset);
+    *payload_ptr = this_offset >> 6;
+    ++payload_ptr;
+    RTC_DCHECK_LE(kPayloadLength, 1023);  // Max length described by 10 bits.
+    *payload_ptr = ((this_offset & 0x3F) << 2) | (kPayloadLength >> 8);
+    ++payload_ptr;
+    *payload_ptr = kPayloadLength & 0xFF;
+    ++payload_ptr;
+  }
+  for (size_t i = 0; i < num_payloads; ++i) {
+    // Write `i` to all bytes in each payload.
+    if (embed_opus_fec) {
+      CreateOpusFecPayload(payload_ptr, kPayloadLength,
+                           static_cast<uint8_t>(i));
+    } else {
+      memset(payload_ptr, static_cast<int>(i), kPayloadLength);
+    }
+    payload_ptr += kPayloadLength;
+  }
+  return packet;
+}
+
+// Create a packet with all payload bytes set to `payload_value`.
+Packet CreatePacket(uint8_t payload_type,
+                    size_t payload_length,
+                    uint8_t payload_value,
+                    bool opus_fec = false) {
+  Packet packet;
+  packet.payload_type = payload_type;
+  packet.timestamp = kBaseTimestamp;
+  packet.sequence_number = kSequenceNumber;
+  packet.payload.SetSize(payload_length);
+  if (opus_fec) {
+    CreateOpusFecPayload(packet.payload.data(), packet.payload.size(),
+                         payload_value);
+  } else {
+    memset(packet.payload.data(), payload_value, packet.payload.size());
+  }
+  return packet;
+}
+
+// Checks that `packet` has the attributes given in the remaining parameters.
+void VerifyPacket(const Packet& packet,
+                  size_t payload_length,
+                  uint8_t payload_type,
+                  uint16_t sequence_number,
+                  uint32_t timestamp,
+                  uint8_t payload_value,
+                  Packet::Priority priority) {
+  EXPECT_EQ(payload_length, packet.payload.size());
+  EXPECT_EQ(payload_type, packet.payload_type);
+  EXPECT_EQ(sequence_number, packet.sequence_number);
+  EXPECT_EQ(timestamp, packet.timestamp);
+  EXPECT_EQ(priority, packet.priority);
+  ASSERT_FALSE(packet.payload.empty());
+  for (size_t i = 0; i < packet.payload.size(); ++i) {
+    ASSERT_EQ(payload_value, packet.payload.data()[i]);
+  }
+}
+
+void VerifyPacket(const Packet& packet,
+                  size_t payload_length,
+                  uint8_t payload_type,
+                  uint16_t sequence_number,
+                  uint32_t timestamp,
+                  uint8_t payload_value,
+                  bool primary) {
+  return VerifyPacket(packet, payload_length, payload_type, sequence_number,
+                      timestamp, payload_value,
+                      Packet::Priority{0, primary ? 0 : 1});
+}
+
+// Start of test definitions.
+
+TEST(RedPayloadSplitter, CreateAndDestroy) {
+  RedPayloadSplitter* splitter = new RedPayloadSplitter;
+  delete splitter;
+}
+
+// Packet A is split into A1 and A2.
+TEST(RedPayloadSplitter, OnePacketTwoPayloads) {
+  uint8_t payload_types[] = {0, 0};
+  const int kTimestampOffset = 160;
+  PacketList packet_list;
+  packet_list.push_back(CreateRedPayload(2, payload_types, kTimestampOffset));
+  RedPayloadSplitter splitter;
+  EXPECT_TRUE(splitter.SplitRed(&packet_list));
+  ASSERT_EQ(2u, packet_list.size());
+  // Check first packet. The first in list should always be the primary payload.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[1],
+               kSequenceNumber, kBaseTimestamp, 1, true);
+  packet_list.pop_front();
+  // Check second packet.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber, kBaseTimestamp - kTimestampOffset, 0, false);
+}
+
+// Packets A and B are not split at all. Only the RED header in each packet is
+// removed.
+TEST(RedPayloadSplitter, TwoPacketsOnePayload) {
+  uint8_t payload_types[] = {0};
+  const int kTimestampOffset = 160;
+  // Create first packet, with a single RED payload.
+  PacketList packet_list;
+  packet_list.push_back(CreateRedPayload(1, payload_types, kTimestampOffset));
+  // Create second packet, with a single RED payload.
+  {
+    Packet packet = CreateRedPayload(1, payload_types, kTimestampOffset);
+    // Manually change timestamp and sequence number of second packet.
+    packet.timestamp += kTimestampOffset;
+    packet.sequence_number++;
+    packet_list.push_back(std::move(packet));
+  }
+  RedPayloadSplitter splitter;
+  EXPECT_TRUE(splitter.SplitRed(&packet_list));
+  ASSERT_EQ(2u, packet_list.size());
+  // Check first packet.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber, kBaseTimestamp, 0, true);
+  packet_list.pop_front();
+  // Check second packet.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber + 1, kBaseTimestamp + kTimestampOffset, 0, true);
+}
+
+// Packets A and B are split into packets A1, A2, A3, B1, B2, B3, with
+// attributes as follows:
+//
+//                  A1*   A2    A3    B1*   B2    B3
+// Payload type     0     1     2     0     1     2
+// Timestamp        b     b-o   b-2o  b+o   b     b-o
+// Sequence number  0     0     0     1     1     1
+//
+// b = kBaseTimestamp, o = kTimestampOffset, * = primary.
+TEST(RedPayloadSplitter, TwoPacketsThreePayloads) {
+  uint8_t payload_types[] = {2, 1, 0};  // Primary is the last one.
+  const int kTimestampOffset = 160;
+  // Create first packet, with 3 RED payloads.
+  PacketList packet_list;
+  packet_list.push_back(CreateRedPayload(3, payload_types, kTimestampOffset));
+  // Create first packet, with 3 RED payloads.
+  {
+    Packet packet = CreateRedPayload(3, payload_types, kTimestampOffset);
+    // Manually change timestamp and sequence number of second packet.
+    packet.timestamp += kTimestampOffset;
+    packet.sequence_number++;
+    packet_list.push_back(std::move(packet));
+  }
+  RedPayloadSplitter splitter;
+  EXPECT_TRUE(splitter.SplitRed(&packet_list));
+  ASSERT_EQ(6u, packet_list.size());
+  // Check first packet, A1.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[2],
+               kSequenceNumber, kBaseTimestamp, 2, {0, 0});
+  packet_list.pop_front();
+  // Check second packet, A2.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[1],
+               kSequenceNumber, kBaseTimestamp - kTimestampOffset, 1, {0, 1});
+  packet_list.pop_front();
+  // Check third packet, A3.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber, kBaseTimestamp - 2 * kTimestampOffset, 0,
+               {0, 2});
+  packet_list.pop_front();
+  // Check fourth packet, B1.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[2],
+               kSequenceNumber + 1, kBaseTimestamp + kTimestampOffset, 2,
+               {0, 0});
+  packet_list.pop_front();
+  // Check fifth packet, B2.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[1],
+               kSequenceNumber + 1, kBaseTimestamp, 1, {0, 1});
+  packet_list.pop_front();
+  // Check sixth packet, B3.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber + 1, kBaseTimestamp - kTimestampOffset, 0,
+               {0, 2});
+}
+
+// Creates a list with 4 packets with these payload types:
+// 0 = CNGnb
+// 1 = PCMu
+// 2 = DTMF (AVT)
+// 3 = iLBC
+// We expect the method CheckRedPayloads to discard the iLBC packet, since it
+// is a non-CNG, non-DTMF payload of another type than the first speech payload
+// found in the list (which is PCMu).
+TEST(RedPayloadSplitter, CheckRedPayloads) {
+  PacketList packet_list;
+  for (uint8_t i = 0; i <= 3; ++i) {
+    // Create packet with payload type `i`, payload length 10 bytes, all 0.
+    packet_list.push_back(CreatePacket(i, 10, 0));
+  }
+
+  // Use a real DecoderDatabase object here instead of a mock, since it is
+  // easier to just register the payload types and let the actual implementation
+  // do its job.
+  DecoderDatabase decoder_database(
+      rtc::make_ref_counted<MockAudioDecoderFactory>(), absl::nullopt);
+  decoder_database.RegisterPayload(0, SdpAudioFormat("cn", 8000, 1));
+  decoder_database.RegisterPayload(1, SdpAudioFormat("pcmu", 8000, 1));
+  decoder_database.RegisterPayload(2,
+                                   SdpAudioFormat("telephone-event", 8000, 1));
+  decoder_database.RegisterPayload(3, SdpAudioFormat("ilbc", 8000, 1));
+
+  RedPayloadSplitter splitter;
+  splitter.CheckRedPayloads(&packet_list, decoder_database);
+
+  ASSERT_EQ(3u, packet_list.size());  // Should have dropped the last packet.
+  // Verify packets. The loop verifies that payload types 0, 1, and 2 are in the
+  // list.
+  for (int i = 0; i <= 2; ++i) {
+    VerifyPacket(packet_list.front(), 10, i, kSequenceNumber, kBaseTimestamp, 0,
+                 true);
+    packet_list.pop_front();
+  }
+  EXPECT_TRUE(packet_list.empty());
+}
+
+// This test creates a RED packet where the payloads also have the payload type
+// for RED. That is, some kind of weird nested RED packet. This is not supported
+// and the splitter should discard all packets.
+TEST(RedPayloadSplitter, CheckRedPayloadsRecursiveRed) {
+  PacketList packet_list;
+  for (uint8_t i = 0; i <= 3; ++i) {
+    // Create packet with RED payload type, payload length 10 bytes, all 0.
+    packet_list.push_back(CreatePacket(kRedPayloadType, 10, 0));
+  }
+
+  // Use a real DecoderDatabase object here instead of a mock, since it is
+  // easier to just register the payload types and let the actual implementation
+  // do its job.
+  DecoderDatabase decoder_database(
+      rtc::make_ref_counted<MockAudioDecoderFactory>(), absl::nullopt);
+  decoder_database.RegisterPayload(kRedPayloadType,
+                                   SdpAudioFormat("red", 8000, 1));
+
+  RedPayloadSplitter splitter;
+  splitter.CheckRedPayloads(&packet_list, decoder_database);
+
+  EXPECT_TRUE(packet_list.empty());  // Should have dropped all packets.
+}
+
+// Packet A is split into A1, A2 and A3. But the length parameter is off, so
+// the last payloads should be discarded.
+TEST(RedPayloadSplitter, WrongPayloadLength) {
+  uint8_t payload_types[] = {0, 0, 0};
+  const int kTimestampOffset = 160;
+  PacketList packet_list;
+  {
+    Packet packet = CreateRedPayload(3, payload_types, kTimestampOffset);
+    // Manually tamper with the payload length of the packet.
+    // This is one byte too short for the second payload (out of three).
+    // We expect only the first payload to be returned.
+    packet.payload.SetSize(packet.payload.size() - (kPayloadLength + 1));
+    packet_list.push_back(std::move(packet));
+  }
+  RedPayloadSplitter splitter;
+  EXPECT_FALSE(splitter.SplitRed(&packet_list));
+  ASSERT_EQ(1u, packet_list.size());
+  // Check first packet.
+  VerifyPacket(packet_list.front(), kPayloadLength, payload_types[0],
+               kSequenceNumber, kBaseTimestamp - 2 * kTimestampOffset, 0,
+               {0, 2});
+  packet_list.pop_front();
+}
+
+// Test that we reject packets too short to contain a RED header.
+TEST(RedPayloadSplitter, RejectsIncompleteHeaders) {
+  RedPayloadSplitter splitter;
+
+  uint8_t payload_types[] = {0, 0};
+  const int kTimestampOffset = 160;
+
+  PacketList packet_list;
+
+  // Truncate the packet such that the first block can not be parsed.
+  packet_list.push_back(CreateRedPayload(2, payload_types, kTimestampOffset));
+  packet_list.front().payload.SetSize(4);
+  EXPECT_FALSE(splitter.SplitRed(&packet_list));
+  EXPECT_FALSE(packet_list.empty());
+
+  // Truncate the packet such that the first block can not be parsed.
+  packet_list.front().payload.SetSize(3);
+  EXPECT_FALSE(splitter.SplitRed(&packet_list));
+  EXPECT_FALSE(packet_list.empty());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.cc
new file mode 100644
index 0000000000..f6e073fc88
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.cc
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/reorder_optimizer.h"
+
+#include <algorithm>
+#include <limits>
+#include <vector>
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDelayBuckets = 100;
+constexpr int kBucketSizeMs = 20;
+
+}  // namespace
+
+ReorderOptimizer::ReorderOptimizer(int forget_factor,
+                                   int ms_per_loss_percent,
+                                   absl::optional<int> start_forget_weight)
+    : histogram_(kDelayBuckets, forget_factor, start_forget_weight),
+      ms_per_loss_percent_(ms_per_loss_percent) {}
+
+void ReorderOptimizer::Update(int relative_delay_ms,
+                              bool reordered,
+                              int base_delay_ms) {
+  const int index = reordered ? relative_delay_ms / kBucketSizeMs : 0;
+  if (index < histogram_.NumBuckets()) {
+    // Maximum delay to register is 2000 ms.
+    histogram_.Add(index);
+  }
+  int bucket_index = MinimizeCostFunction(base_delay_ms);
+  optimal_delay_ms_ = (1 + bucket_index) * kBucketSizeMs;
+}
+
+void ReorderOptimizer::Reset() {
+  histogram_.Reset();
+  optimal_delay_ms_.reset();
+}
+
+int ReorderOptimizer::MinimizeCostFunction(int base_delay_ms) const {
+  const std::vector<int>& buckets = histogram_.buckets();
+
+  // Values are calculated in Q30.
+  int64_t loss_probability = 1 << 30;
+  int64_t min_cost = std::numeric_limits<int64_t>::max();
+  int min_bucket = 0;
+  for (int i = 0; i < static_cast<int>(buckets.size()); ++i) {
+    loss_probability -= buckets[i];
+    int64_t delay_ms =
+        static_cast<int64_t>(std::max(0, i * kBucketSizeMs - base_delay_ms))
+        << 30;
+    int64_t cost = delay_ms + 100 * ms_per_loss_percent_ * loss_probability;
+
+    if (cost < min_cost) {
+      min_cost = cost;
+      min_bucket = i;
+    }
+    if (loss_probability == 0) {
+      break;
+    }
+  }
+
+  return min_bucket;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.h b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.h
new file mode 100644
index 0000000000..06f6bc7e50
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_REORDER_OPTIMIZER_H_
+#define MODULES_AUDIO_CODING_NETEQ_REORDER_OPTIMIZER_H_
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/histogram.h"
+
+namespace webrtc {
+
+// Calculates an optimal delay to reduce the chance of missing reordered
+// packets. The delay/loss trade-off can be tune using the `ms_per_loss_percent`
+// parameter.
+class ReorderOptimizer {
+ public:
+  ReorderOptimizer(int forget_factor,
+                   int ms_per_loss_percent,
+                   absl::optional<int> start_forget_weight);
+
+  void Update(int relative_delay_ms, bool reordered, int base_delay_ms);
+
+  absl::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }
+
+  void Reset();
+
+ private:
+  int MinimizeCostFunction(int base_delay_ms) const;
+
+  Histogram histogram_;
+  const int ms_per_loss_percent_;
+  absl::optional<int> optimal_delay_ms_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_REORDER_OPTIMIZER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer_unittest.cc
new file mode 100644
index 0000000000..aaa1062560
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/reorder_optimizer_unittest.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/reorder_optimizer.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kForgetFactor = 32745;  // 0.9993 in Q15.
+constexpr int kMsPerLossPercent = 20;
+constexpr int kStartForgetWeight = 1;
+
+}  // namespace
+
+TEST(ReorderOptimizerTest, OnlyIncreaseDelayForReorderedPackets) {
+  ReorderOptimizer reorder_optimizer(kForgetFactor, kMsPerLossPercent,
+                                     kStartForgetWeight);
+  EXPECT_FALSE(reorder_optimizer.GetOptimalDelayMs());
+
+  // Delay should not increase for in-order packets.
+  reorder_optimizer.Update(60, /*reordered=*/false, 0);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 20);
+
+  reorder_optimizer.Update(100, /*reordered=*/false, 0);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 20);
+
+  reorder_optimizer.Update(80, /*reordered=*/true, 0);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 100);
+}
+
+TEST(ReorderOptimizerTest, AvoidIncreasingDelayWhenProbabilityIsLow) {
+  ReorderOptimizer reorder_optimizer(kForgetFactor, kMsPerLossPercent,
+                                     kStartForgetWeight);
+
+  reorder_optimizer.Update(40, /*reordered=*/true, 0);
+  reorder_optimizer.Update(40, /*reordered=*/true, 0);
+  reorder_optimizer.Update(40, /*reordered=*/true, 0);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 60);
+
+  // The cost of the delay is too high relative the probability.
+  reorder_optimizer.Update(600, /*reordered=*/true, 0);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 60);
+}
+
+TEST(ReorderOptimizerTest, BaseDelayIsSubtractedFromCost) {
+  constexpr int kBaseDelayMs = 200;
+  ReorderOptimizer reorder_optimizer(kForgetFactor, kMsPerLossPercent,
+                                     kStartForgetWeight);
+
+  reorder_optimizer.Update(40, /*reordered=*/true, kBaseDelayMs);
+  reorder_optimizer.Update(40, /*reordered=*/true, kBaseDelayMs);
+  reorder_optimizer.Update(40, /*reordered=*/true, kBaseDelayMs);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 60);
+
+  // The cost of the delay is too high relative the probability.
+  reorder_optimizer.Update(600, /*reordered=*/true, kBaseDelayMs);
+  EXPECT_EQ(reorder_optimizer.GetOptimalDelayMs(), 620);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.cc b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.cc
new file mode 100644
index 0000000000..52d3fa90f1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.cc
@@ -0,0 +1,394 @@
+/*
+ *  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 "modules/audio_coding/neteq/statistics_calculator.h"
+
+#include <string.h>  // memset
+
+#include <algorithm>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+size_t AddIntToSizeTWithLowerCap(int a, size_t b) {
+  const size_t ret = b + a;
+  // If a + b is negative, resulting in a negative wrap, cap it to zero instead.
+  static_assert(sizeof(size_t) >= sizeof(int),
+                "int must not be wider than size_t for this to work");
+  return (a < 0 && ret > b) ? 0 : ret;
+}
+
+constexpr int kInterruptionLenMs = 150;
+}  // namespace
+
+// Allocating the static const so that it can be passed by reference to
+// RTC_DCHECK.
+const size_t StatisticsCalculator::kLenWaitingTimes;
+
+StatisticsCalculator::PeriodicUmaLogger::PeriodicUmaLogger(
+    absl::string_view uma_name,
+    int report_interval_ms,
+    int max_value)
+    : uma_name_(uma_name),
+      report_interval_ms_(report_interval_ms),
+      max_value_(max_value),
+      timer_(0) {}
+
+StatisticsCalculator::PeriodicUmaLogger::~PeriodicUmaLogger() = default;
+
+void StatisticsCalculator::PeriodicUmaLogger::AdvanceClock(int step_ms) {
+  timer_ += step_ms;
+  if (timer_ < report_interval_ms_) {
+    return;
+  }
+  LogToUma(Metric());
+  Reset();
+  timer_ -= report_interval_ms_;
+  RTC_DCHECK_GE(timer_, 0);
+}
+
+void StatisticsCalculator::PeriodicUmaLogger::LogToUma(int value) const {
+  RTC_HISTOGRAM_COUNTS_SPARSE(uma_name_, value, 1, max_value_, 50);
+}
+
+StatisticsCalculator::PeriodicUmaCount::PeriodicUmaCount(
+    absl::string_view uma_name,
+    int report_interval_ms,
+    int max_value)
+    : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {}
+
+StatisticsCalculator::PeriodicUmaCount::~PeriodicUmaCount() {
+  // Log the count for the current (incomplete) interval.
+  LogToUma(Metric());
+}
+
+void StatisticsCalculator::PeriodicUmaCount::RegisterSample() {
+  ++counter_;
+}
+
+int StatisticsCalculator::PeriodicUmaCount::Metric() const {
+  return counter_;
+}
+
+void StatisticsCalculator::PeriodicUmaCount::Reset() {
+  counter_ = 0;
+}
+
+StatisticsCalculator::PeriodicUmaAverage::PeriodicUmaAverage(
+    absl::string_view uma_name,
+    int report_interval_ms,
+    int max_value)
+    : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {}
+
+StatisticsCalculator::PeriodicUmaAverage::~PeriodicUmaAverage() {
+  // Log the average for the current (incomplete) interval.
+  LogToUma(Metric());
+}
+
+void StatisticsCalculator::PeriodicUmaAverage::RegisterSample(int value) {
+  sum_ += value;
+  ++counter_;
+}
+
+int StatisticsCalculator::PeriodicUmaAverage::Metric() const {
+  return counter_ == 0 ? 0 : static_cast<int>(sum_ / counter_);
+}
+
+void StatisticsCalculator::PeriodicUmaAverage::Reset() {
+  sum_ = 0.0;
+  counter_ = 0;
+}
+
+StatisticsCalculator::StatisticsCalculator()
+    : preemptive_samples_(0),
+      accelerate_samples_(0),
+      expanded_speech_samples_(0),
+      expanded_noise_samples_(0),
+      timestamps_since_last_report_(0),
+      secondary_decoded_samples_(0),
+      discarded_secondary_packets_(0),
+      delayed_packet_outage_counter_(
+          "WebRTC.Audio.DelayedPacketOutageEventsPerMinute",
+          60000,  // 60 seconds report interval.
+          100),
+      excess_buffer_delay_("WebRTC.Audio.AverageExcessBufferDelayMs",
+                           60000,  // 60 seconds report interval.
+                           1000),
+      buffer_full_counter_("WebRTC.Audio.JitterBufferFullPerMinute",
+                           60000,  // 60 seconds report interval.
+                           100) {}
+
+StatisticsCalculator::~StatisticsCalculator() = default;
+
+void StatisticsCalculator::Reset() {
+  preemptive_samples_ = 0;
+  accelerate_samples_ = 0;
+  expanded_speech_samples_ = 0;
+  expanded_noise_samples_ = 0;
+  secondary_decoded_samples_ = 0;
+  discarded_secondary_packets_ = 0;
+  waiting_times_.clear();
+}
+
+void StatisticsCalculator::ResetMcu() {
+  timestamps_since_last_report_ = 0;
+}
+
+void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples,
+                                                bool is_new_concealment_event) {
+  expanded_speech_samples_ += num_samples;
+  ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), true);
+  lifetime_stats_.concealment_events += is_new_concealment_event;
+}
+
+void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples,
+                                                bool is_new_concealment_event) {
+  expanded_noise_samples_ += num_samples;
+  ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), false);
+  lifetime_stats_.concealment_events += is_new_concealment_event;
+}
+
+void StatisticsCalculator::ExpandedVoiceSamplesCorrection(int num_samples) {
+  expanded_speech_samples_ =
+      AddIntToSizeTWithLowerCap(num_samples, expanded_speech_samples_);
+  ConcealedSamplesCorrection(num_samples, true);
+}
+
+void StatisticsCalculator::ExpandedNoiseSamplesCorrection(int num_samples) {
+  expanded_noise_samples_ =
+      AddIntToSizeTWithLowerCap(num_samples, expanded_noise_samples_);
+  ConcealedSamplesCorrection(num_samples, false);
+}
+
+void StatisticsCalculator::DecodedOutputPlayed() {
+  decoded_output_played_ = true;
+}
+
+void StatisticsCalculator::EndExpandEvent(int fs_hz) {
+  RTC_DCHECK_GE(lifetime_stats_.concealed_samples,
+                concealed_samples_at_event_end_);
+  const int event_duration_ms =
+      1000 *
+      (lifetime_stats_.concealed_samples - concealed_samples_at_event_end_) /
+      fs_hz;
+  if (event_duration_ms >= kInterruptionLenMs && decoded_output_played_) {
+    lifetime_stats_.interruption_count++;
+    lifetime_stats_.total_interruption_duration_ms += event_duration_ms;
+    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.AudioInterruptionMs", event_duration_ms,
+                         /*min=*/150, /*max=*/5000, /*bucket_count=*/50);
+  }
+  concealed_samples_at_event_end_ = lifetime_stats_.concealed_samples;
+}
+
+void StatisticsCalculator::ConcealedSamplesCorrection(int num_samples,
+                                                      bool is_voice) {
+  if (num_samples < 0) {
+    // Store negative correction to subtract from future positive additions.
+    // See also the function comment in the header file.
+    concealed_samples_correction_ -= num_samples;
+    if (!is_voice) {
+      silent_concealed_samples_correction_ -= num_samples;
+    }
+    return;
+  }
+
+  const size_t canceled_out =
+      std::min(static_cast<size_t>(num_samples), concealed_samples_correction_);
+  concealed_samples_correction_ -= canceled_out;
+  lifetime_stats_.concealed_samples += num_samples - canceled_out;
+
+  if (!is_voice) {
+    const size_t silent_canceled_out = std::min(
+        static_cast<size_t>(num_samples), silent_concealed_samples_correction_);
+    silent_concealed_samples_correction_ -= silent_canceled_out;
+    lifetime_stats_.silent_concealed_samples +=
+        num_samples - silent_canceled_out;
+  }
+}
+
+void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) {
+  preemptive_samples_ += num_samples;
+  operations_and_state_.preemptive_samples += num_samples;
+  lifetime_stats_.inserted_samples_for_deceleration += num_samples;
+}
+
+void StatisticsCalculator::AcceleratedSamples(size_t num_samples) {
+  accelerate_samples_ += num_samples;
+  operations_and_state_.accelerate_samples += num_samples;
+  lifetime_stats_.removed_samples_for_acceleration += num_samples;
+}
+
+void StatisticsCalculator::GeneratedNoiseSamples(size_t num_samples) {
+  lifetime_stats_.generated_noise_samples += num_samples;
+}
+
+void StatisticsCalculator::PacketsDiscarded(size_t num_packets) {
+  lifetime_stats_.packets_discarded += num_packets;
+}
+
+void StatisticsCalculator::SecondaryPacketsDiscarded(size_t num_packets) {
+  discarded_secondary_packets_ += num_packets;
+  lifetime_stats_.fec_packets_discarded += num_packets;
+}
+
+void StatisticsCalculator::SecondaryPacketsReceived(size_t num_packets) {
+  lifetime_stats_.fec_packets_received += num_packets;
+}
+
+void StatisticsCalculator::IncreaseCounter(size_t num_samples, int fs_hz) {
+  const int time_step_ms =
+      rtc::CheckedDivExact(static_cast<int>(1000 * num_samples), fs_hz);
+  delayed_packet_outage_counter_.AdvanceClock(time_step_ms);
+  excess_buffer_delay_.AdvanceClock(time_step_ms);
+  buffer_full_counter_.AdvanceClock(time_step_ms);
+  timestamps_since_last_report_ += static_cast<uint32_t>(num_samples);
+  if (timestamps_since_last_report_ >
+      static_cast<uint32_t>(fs_hz * kMaxReportPeriod)) {
+    timestamps_since_last_report_ = 0;
+  }
+  lifetime_stats_.total_samples_received += num_samples;
+}
+
+void StatisticsCalculator::JitterBufferDelay(
+    size_t num_samples,
+    uint64_t waiting_time_ms,
+    uint64_t target_delay_ms,
+    uint64_t unlimited_target_delay_ms) {
+  lifetime_stats_.jitter_buffer_delay_ms += waiting_time_ms * num_samples;
+  lifetime_stats_.jitter_buffer_target_delay_ms +=
+      target_delay_ms * num_samples;
+  lifetime_stats_.jitter_buffer_minimum_delay_ms +=
+      unlimited_target_delay_ms * num_samples;
+  lifetime_stats_.jitter_buffer_emitted_count += num_samples;
+}
+
+void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
+  secondary_decoded_samples_ += num_samples;
+}
+
+void StatisticsCalculator::FlushedPacketBuffer() {
+  operations_and_state_.packet_buffer_flushes++;
+  buffer_full_counter_.RegisterSample();
+}
+
+void StatisticsCalculator::ReceivedPacket() {
+  ++lifetime_stats_.jitter_buffer_packets_received;
+}
+
+void StatisticsCalculator::RelativePacketArrivalDelay(size_t delay_ms) {
+  lifetime_stats_.relative_packet_arrival_delay_ms += delay_ms;
+}
+
+void StatisticsCalculator::LogDelayedPacketOutageEvent(int num_samples,
+                                                       int fs_hz) {
+  int outage_duration_ms = num_samples / (fs_hz / 1000);
+  RTC_HISTOGRAM_COUNTS("WebRTC.Audio.DelayedPacketOutageEventMs",
+                       outage_duration_ms, 1 /* min */, 2000 /* max */,
+                       100 /* bucket count */);
+  delayed_packet_outage_counter_.RegisterSample();
+  lifetime_stats_.delayed_packet_outage_samples += num_samples;
+}
+
+void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
+  excess_buffer_delay_.RegisterSample(waiting_time_ms);
+  RTC_DCHECK_LE(waiting_times_.size(), kLenWaitingTimes);
+  if (waiting_times_.size() == kLenWaitingTimes) {
+    // Erase first value.
+    waiting_times_.pop_front();
+  }
+  waiting_times_.push_back(waiting_time_ms);
+  operations_and_state_.last_waiting_time_ms = waiting_time_ms;
+}
+
+void StatisticsCalculator::GetNetworkStatistics(size_t samples_per_packet,
+                                                NetEqNetworkStatistics* stats) {
+  RTC_DCHECK(stats);
+
+  stats->accelerate_rate =
+      CalculateQ14Ratio(accelerate_samples_, timestamps_since_last_report_);
+
+  stats->preemptive_rate =
+      CalculateQ14Ratio(preemptive_samples_, timestamps_since_last_report_);
+
+  stats->expand_rate =
+      CalculateQ14Ratio(expanded_speech_samples_ + expanded_noise_samples_,
+                        timestamps_since_last_report_);
+
+  stats->speech_expand_rate = CalculateQ14Ratio(expanded_speech_samples_,
+                                                timestamps_since_last_report_);
+
+  stats->secondary_decoded_rate = CalculateQ14Ratio(
+      secondary_decoded_samples_, timestamps_since_last_report_);
+
+  const size_t discarded_secondary_samples =
+      discarded_secondary_packets_ * samples_per_packet;
+  stats->secondary_discarded_rate =
+      CalculateQ14Ratio(discarded_secondary_samples,
+                        static_cast<uint32_t>(discarded_secondary_samples +
+                                              secondary_decoded_samples_));
+
+  if (waiting_times_.size() == 0) {
+    stats->mean_waiting_time_ms = -1;
+    stats->median_waiting_time_ms = -1;
+    stats->min_waiting_time_ms = -1;
+    stats->max_waiting_time_ms = -1;
+  } else {
+    std::sort(waiting_times_.begin(), waiting_times_.end());
+    // Find mid-point elements. If the size is odd, the two values
+    // `middle_left` and `middle_right` will both be the one middle element; if
+    // the size is even, they will be the the two neighboring elements at the
+    // middle of the list.
+    const int middle_left = waiting_times_[(waiting_times_.size() - 1) / 2];
+    const int middle_right = waiting_times_[waiting_times_.size() / 2];
+    // Calculate the average of the two. (Works also for odd sizes.)
+    stats->median_waiting_time_ms = (middle_left + middle_right) / 2;
+    stats->min_waiting_time_ms = waiting_times_.front();
+    stats->max_waiting_time_ms = waiting_times_.back();
+    double sum = 0;
+    for (auto time : waiting_times_) {
+      sum += time;
+    }
+    stats->mean_waiting_time_ms = static_cast<int>(sum / waiting_times_.size());
+  }
+
+  // Reset counters.
+  ResetMcu();
+  Reset();
+}
+
+NetEqLifetimeStatistics StatisticsCalculator::GetLifetimeStatistics() const {
+  return lifetime_stats_;
+}
+
+NetEqOperationsAndState StatisticsCalculator::GetOperationsAndState() const {
+  return operations_and_state_;
+}
+
+uint16_t StatisticsCalculator::CalculateQ14Ratio(size_t numerator,
+                                                 uint32_t denominator) {
+  if (numerator == 0) {
+    return 0;
+  } else if (numerator < denominator) {
+    // Ratio must be smaller than 1 in Q14.
+    RTC_DCHECK_LT((numerator << 14) / denominator, (1 << 14));
+    return static_cast<uint16_t>((numerator << 14) / denominator);
+  } else {
+    // Will not produce a ratio larger than 1, since this is probably an error.
+    return 1 << 14;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.h b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.h
new file mode 100644
index 0000000000..33a22d02dd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator.h
@@ -0,0 +1,210 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
+
+#include <deque>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/neteq/neteq.h"
+
+namespace webrtc {
+
+class DelayManager;
+
+// This class handles various network statistics in NetEq.
+class StatisticsCalculator {
+ public:
+  StatisticsCalculator();
+
+  virtual ~StatisticsCalculator();
+
+  StatisticsCalculator(const StatisticsCalculator&) = delete;
+  StatisticsCalculator& operator=(const StatisticsCalculator&) = delete;
+
+  // Resets most of the counters.
+  void Reset();
+
+  // Resets the counters that are not handled by Reset().
+  void ResetMcu();
+
+  // Reports that `num_samples` samples were produced through expansion, and
+  // that the expansion produced other than just noise samples.
+  void ExpandedVoiceSamples(size_t num_samples, bool is_new_concealment_event);
+
+  // Reports that `num_samples` samples were produced through expansion, and
+  // that the expansion produced only noise samples.
+  void ExpandedNoiseSamples(size_t num_samples, bool is_new_concealment_event);
+
+  // Corrects the statistics for number of samples produced through non-noise
+  // expansion by adding `num_samples` (negative or positive) to the current
+  // value. The result is capped to zero to avoid negative values.
+  void ExpandedVoiceSamplesCorrection(int num_samples);
+
+  // Same as ExpandedVoiceSamplesCorrection but for noise samples.
+  void ExpandedNoiseSamplesCorrection(int num_samples);
+
+  void DecodedOutputPlayed();
+
+  // Mark end of expand event; triggers some stats to be reported.
+  void EndExpandEvent(int fs_hz);
+
+  // Reports that `num_samples` samples were produced through preemptive
+  // expansion.
+  void PreemptiveExpandedSamples(size_t num_samples);
+
+  // Reports that `num_samples` samples were removed through accelerate.
+  void AcceleratedSamples(size_t num_samples);
+
+  // Reports that `num_samples` comfort noise samples were generated.
+  void GeneratedNoiseSamples(size_t num_samples);
+
+  // Reports that `num_packets` packets were discarded.
+  virtual void PacketsDiscarded(size_t num_packets);
+
+  // Reports that `num_packets` secondary (FEC) packets were discarded.
+  virtual void SecondaryPacketsDiscarded(size_t num_packets);
+
+  // Reports that `num_packets` secondary (FEC) packets were received.
+  virtual void SecondaryPacketsReceived(size_t num_packets);
+
+  // Increases the report interval counter with `num_samples` at a sample rate
+  // of `fs_hz`. This is how the StatisticsCalculator gets notified that current
+  // time is increasing.
+  void IncreaseCounter(size_t num_samples, int fs_hz);
+
+  // Update jitter buffer delay counter.
+  void JitterBufferDelay(size_t num_samples,
+                         uint64_t waiting_time_ms,
+                         uint64_t target_delay_ms,
+                         uint64_t unlimited_target_delay_ms);
+
+  // Stores new packet waiting time in waiting time statistics.
+  void StoreWaitingTime(int waiting_time_ms);
+
+  // Reports that `num_samples` samples were decoded from secondary packets.
+  void SecondaryDecodedSamples(int num_samples);
+
+  // Reports that the packet buffer was flushed.
+  void FlushedPacketBuffer();
+
+  // Reports that the jitter buffer received a packet.
+  void ReceivedPacket();
+
+  // Reports that a received packet was delayed by `delay_ms` milliseconds.
+  virtual void RelativePacketArrivalDelay(size_t delay_ms);
+
+  // Logs a delayed packet outage event of `num_samples` expanded at a sample
+  // rate of `fs_hz`. A delayed packet outage event is defined as an expand
+  // period caused not by an actual packet loss, but by a delayed packet.
+  virtual void LogDelayedPacketOutageEvent(int num_samples, int fs_hz);
+
+  // Returns the current network statistics in `stats`. The number of samples
+  // per packet is `samples_per_packet`. The method does not populate
+  // `preferred_buffer_size_ms`, `jitter_peaks_found` or `clockdrift_ppm`; use
+  // the PopulateDelayManagerStats method for those.
+  void GetNetworkStatistics(size_t samples_per_packet,
+                            NetEqNetworkStatistics* stats);
+
+  // Returns a copy of this class's lifetime statistics. These statistics are
+  // never reset.
+  NetEqLifetimeStatistics GetLifetimeStatistics() const;
+
+  NetEqOperationsAndState GetOperationsAndState() const;
+
+ private:
+  static const int kMaxReportPeriod = 60;  // Seconds before auto-reset.
+  static const size_t kLenWaitingTimes = 100;
+
+  class PeriodicUmaLogger {
+   public:
+    PeriodicUmaLogger(absl::string_view uma_name,
+                      int report_interval_ms,
+                      int max_value);
+    virtual ~PeriodicUmaLogger();
+    void AdvanceClock(int step_ms);
+
+   protected:
+    void LogToUma(int value) const;
+    virtual int Metric() const = 0;
+    virtual void Reset() = 0;
+
+    const std::string uma_name_;
+    const int report_interval_ms_;
+    const int max_value_;
+    int timer_ = 0;
+  };
+
+  class PeriodicUmaCount final : public PeriodicUmaLogger {
+   public:
+    PeriodicUmaCount(absl::string_view uma_name,
+                     int report_interval_ms,
+                     int max_value);
+    ~PeriodicUmaCount() override;
+    void RegisterSample();
+
+   protected:
+    int Metric() const override;
+    void Reset() override;
+
+   private:
+    int counter_ = 0;
+  };
+
+  class PeriodicUmaAverage final : public PeriodicUmaLogger {
+   public:
+    PeriodicUmaAverage(absl::string_view uma_name,
+                       int report_interval_ms,
+                       int max_value);
+    ~PeriodicUmaAverage() override;
+    void RegisterSample(int value);
+
+   protected:
+    int Metric() const override;
+    void Reset() override;
+
+   private:
+    double sum_ = 0.0;
+    int counter_ = 0;
+  };
+
+  // Corrects the concealed samples counter in lifetime_stats_. The value of
+  // num_samples_ is added directly to the stat if the correction is positive.
+  // If the correction is negative, it is cached and will be subtracted against
+  // future additions to the counter. This is meant to be called from
+  // Expanded{Voice,Noise}Samples{Correction}.
+  void ConcealedSamplesCorrection(int num_samples, bool is_voice);
+
+  // Calculates numerator / denominator, and returns the value in Q14.
+  static uint16_t CalculateQ14Ratio(size_t numerator, uint32_t denominator);
+
+  NetEqLifetimeStatistics lifetime_stats_;
+  NetEqOperationsAndState operations_and_state_;
+  size_t concealed_samples_correction_ = 0;
+  size_t silent_concealed_samples_correction_ = 0;
+  size_t preemptive_samples_;
+  size_t accelerate_samples_;
+  size_t expanded_speech_samples_;
+  size_t expanded_noise_samples_;
+  size_t concealed_samples_at_event_end_ = 0;
+  uint32_t timestamps_since_last_report_;
+  std::deque<int> waiting_times_;
+  uint32_t secondary_decoded_samples_;
+  size_t discarded_secondary_packets_;
+  PeriodicUmaCount delayed_packet_outage_counter_;
+  PeriodicUmaAverage excess_buffer_delay_;
+  PeriodicUmaCount buffer_full_counter_;
+  bool decoded_output_played_ = false;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator_unittest.cc
new file mode 100644
index 0000000000..491cd83dc4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/statistics_calculator_unittest.cc
@@ -0,0 +1,206 @@
+/*
+ *  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/neteq/statistics_calculator.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(LifetimeStatistics, TotalSamplesReceived) {
+  StatisticsCalculator stats;
+  for (int i = 0; i < 10; ++i) {
+    stats.IncreaseCounter(480, 48000);  // 10 ms at 48 kHz.
+  }
+  EXPECT_EQ(10 * 480u, stats.GetLifetimeStatistics().total_samples_received);
+}
+
+TEST(LifetimeStatistics, SamplesConcealed) {
+  StatisticsCalculator stats;
+  stats.ExpandedVoiceSamples(100, false);
+  stats.ExpandedNoiseSamples(17, false);
+  EXPECT_EQ(100u + 17u, stats.GetLifetimeStatistics().concealed_samples);
+}
+
+// This test verifies that a negative correction of concealed_samples does not
+// result in a decrease in the stats value (because stats-consuming applications
+// would not expect the value to decrease). Instead, the correction should be
+// made to future increments to the stat.
+TEST(LifetimeStatistics, SamplesConcealedCorrection) {
+  StatisticsCalculator stats;
+  stats.ExpandedVoiceSamples(100, false);
+  EXPECT_EQ(100u, stats.GetLifetimeStatistics().concealed_samples);
+  stats.ExpandedVoiceSamplesCorrection(-10);
+  // Do not subtract directly, but keep the correction for later.
+  EXPECT_EQ(100u, stats.GetLifetimeStatistics().concealed_samples);
+  stats.ExpandedVoiceSamplesCorrection(20);
+  // The total correction is 20 - 10.
+  EXPECT_EQ(110u, stats.GetLifetimeStatistics().concealed_samples);
+
+  // Also test correction done to the next ExpandedVoiceSamples call.
+  stats.ExpandedVoiceSamplesCorrection(-17);
+  EXPECT_EQ(110u, stats.GetLifetimeStatistics().concealed_samples);
+  stats.ExpandedVoiceSamples(100, false);
+  EXPECT_EQ(110u + 100u - 17u, stats.GetLifetimeStatistics().concealed_samples);
+}
+
+// This test verifies that neither "accelerate" nor "pre-emptive expand" reults
+// in a modification to concealed_samples stats. Only PLC operations (i.e.,
+// "expand" and "merge") should affect the stat.
+TEST(LifetimeStatistics, NoUpdateOnTimeStretch) {
+  StatisticsCalculator stats;
+  stats.ExpandedVoiceSamples(100, false);
+  stats.AcceleratedSamples(4711);
+  stats.PreemptiveExpandedSamples(17);
+  stats.ExpandedVoiceSamples(100, false);
+  EXPECT_EQ(200u, stats.GetLifetimeStatistics().concealed_samples);
+}
+
+TEST(StatisticsCalculator, ExpandedSamplesCorrection) {
+  StatisticsCalculator stats;
+  NetEqNetworkStatistics stats_output;
+  constexpr int kSampleRateHz = 48000;
+  constexpr int k10MsSamples = kSampleRateHz / 100;
+  constexpr int kPacketSizeMs = 20;
+  constexpr size_t kSamplesPerPacket = kPacketSizeMs * kSampleRateHz / 1000;
+
+  // Advance time by 10 ms.
+  stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
+
+  stats.GetNetworkStatistics(kSamplesPerPacket, &stats_output);
+
+  EXPECT_EQ(0u, stats_output.expand_rate);
+  EXPECT_EQ(0u, stats_output.speech_expand_rate);
+
+  // Correct with a negative value.
+  stats.ExpandedVoiceSamplesCorrection(-100);
+  stats.ExpandedNoiseSamplesCorrection(-100);
+  stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
+  stats.GetNetworkStatistics(kSamplesPerPacket, &stats_output);
+  // Expect no change, since negative values are disallowed.
+  EXPECT_EQ(0u, stats_output.expand_rate);
+  EXPECT_EQ(0u, stats_output.speech_expand_rate);
+
+  // Correct with a positive value.
+  stats.ExpandedVoiceSamplesCorrection(50);
+  stats.ExpandedNoiseSamplesCorrection(200);
+  stats.IncreaseCounter(k10MsSamples, kSampleRateHz);
+  stats.GetNetworkStatistics(kSamplesPerPacket, &stats_output);
+  // Calculate expected rates in Q14. Expand rate is noise + voice, while
+  // speech expand rate is only voice.
+  EXPECT_EQ(((50u + 200u) << 14) / k10MsSamples, stats_output.expand_rate);
+  EXPECT_EQ((50u << 14) / k10MsSamples, stats_output.speech_expand_rate);
+}
+
+TEST(StatisticsCalculator, RelativePacketArrivalDelay) {
+  StatisticsCalculator stats;
+
+  stats.RelativePacketArrivalDelay(50);
+  NetEqLifetimeStatistics stats_output = stats.GetLifetimeStatistics();
+  EXPECT_EQ(50u, stats_output.relative_packet_arrival_delay_ms);
+
+  stats.RelativePacketArrivalDelay(20);
+  stats_output = stats.GetLifetimeStatistics();
+  EXPECT_EQ(70u, stats_output.relative_packet_arrival_delay_ms);
+}
+
+TEST(StatisticsCalculator, ReceivedPacket) {
+  StatisticsCalculator stats;
+
+  stats.ReceivedPacket();
+  NetEqLifetimeStatistics stats_output = stats.GetLifetimeStatistics();
+  EXPECT_EQ(1u, stats_output.jitter_buffer_packets_received);
+
+  stats.ReceivedPacket();
+  stats_output = stats.GetLifetimeStatistics();
+  EXPECT_EQ(2u, stats_output.jitter_buffer_packets_received);
+}
+
+TEST(StatisticsCalculator, InterruptionCounter) {
+  constexpr int fs_khz = 48;
+  constexpr int fs_hz = fs_khz * 1000;
+  StatisticsCalculator stats;
+  stats.DecodedOutputPlayed();
+  stats.EndExpandEvent(fs_hz);
+  auto lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(0, lts.interruption_count);
+  EXPECT_EQ(0, lts.total_interruption_duration_ms);
+
+  // Add an event that is shorter than 150 ms. Should not be logged.
+  stats.ExpandedVoiceSamples(10 * fs_khz, false);   // 10 ms.
+  stats.ExpandedNoiseSamples(139 * fs_khz, false);  // 139 ms.
+  stats.EndExpandEvent(fs_hz);
+  lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(0, lts.interruption_count);
+
+  // Add an event that is longer than 150 ms. Should be logged.
+  stats.ExpandedVoiceSamples(140 * fs_khz, false);  // 140 ms.
+  stats.ExpandedNoiseSamples(11 * fs_khz, false);   // 11 ms.
+  stats.EndExpandEvent(fs_hz);
+  lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(1, lts.interruption_count);
+  EXPECT_EQ(151, lts.total_interruption_duration_ms);
+
+  // Add one more long event.
+  stats.ExpandedVoiceSamples(100 * fs_khz, false);   // 100 ms.
+  stats.ExpandedNoiseSamples(5000 * fs_khz, false);  // 5000 ms.
+  stats.EndExpandEvent(fs_hz);
+  lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(2, lts.interruption_count);
+  EXPECT_EQ(5100 + 151, lts.total_interruption_duration_ms);
+}
+
+TEST(StatisticsCalculator, InterruptionCounterDoNotLogBeforeDecoding) {
+  constexpr int fs_khz = 48;
+  constexpr int fs_hz = fs_khz * 1000;
+  StatisticsCalculator stats;
+
+  // Add an event that is longer than 150 ms. Should normally be logged, but we
+  // have not called DecodedOutputPlayed() yet, so it shouldn't this time.
+  stats.ExpandedVoiceSamples(151 * fs_khz, false);  // 151 ms.
+  stats.EndExpandEvent(fs_hz);
+  auto lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(0, lts.interruption_count);
+
+  // Call DecodedOutputPlayed(). Logging should happen after this.
+  stats.DecodedOutputPlayed();
+
+  // Add one more long event.
+  stats.ExpandedVoiceSamples(151 * fs_khz, false);  // 151 ms.
+  stats.EndExpandEvent(fs_hz);
+  lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(1, lts.interruption_count);
+}
+
+TEST(StatisticsCalculator, DiscardedPackets) {
+  StatisticsCalculator statistics_calculator;
+  EXPECT_EQ(0u,
+            statistics_calculator.GetLifetimeStatistics().packets_discarded);
+
+  statistics_calculator.PacketsDiscarded(1);
+  EXPECT_EQ(1u,
+            statistics_calculator.GetLifetimeStatistics().packets_discarded);
+
+  statistics_calculator.PacketsDiscarded(10);
+  EXPECT_EQ(11u,
+            statistics_calculator.GetLifetimeStatistics().packets_discarded);
+
+  // Calling `SecondaryPacketsDiscarded` does not modify `packets_discarded`.
+  statistics_calculator.SecondaryPacketsDiscarded(1);
+  EXPECT_EQ(11u,
+            statistics_calculator.GetLifetimeStatistics().packets_discarded);
+
+  // Calling `FlushedPacketBuffer` does not modify `packets_discarded`.
+  statistics_calculator.FlushedPacketBuffer();
+  EXPECT_EQ(11u,
+            statistics_calculator.GetLifetimeStatistics().packets_discarded);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.cc
new file mode 100644
index 0000000000..7d7cac7157
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.cc
@@ -0,0 +1,118 @@
+/*
+ *  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/neteq/sync_buffer.h"
+
+#include <algorithm>  // Access to min.
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+size_t SyncBuffer::FutureLength() const {
+  return Size() - next_index_;
+}
+
+void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
+  size_t samples_added = append_this.Size();
+  AudioMultiVector::PushBack(append_this);
+  AudioMultiVector::PopFront(samples_added);
+  if (samples_added <= next_index_) {
+    next_index_ -= samples_added;
+  } else {
+    // This means that we are pushing out future data that was never used.
+    //    RTC_DCHECK_NOTREACHED();
+    // TODO(hlundin): This assert must be disabled to support 60 ms frames.
+    // This should not happen even for 60 ms frames, but it does. Investigate
+    // why.
+    next_index_ = 0;
+  }
+  dtmf_index_ -= std::min(dtmf_index_, samples_added);
+}
+
+void SyncBuffer::PushBackInterleaved(const rtc::BufferT<int16_t>& append_this) {
+  const size_t size_before_adding = Size();
+  AudioMultiVector::PushBackInterleaved(append_this);
+  const size_t samples_added_per_channel = Size() - size_before_adding;
+  RTC_DCHECK_EQ(samples_added_per_channel * Channels(), append_this.size());
+  AudioMultiVector::PopFront(samples_added_per_channel);
+  next_index_ -= std::min(next_index_, samples_added_per_channel);
+  dtmf_index_ -= std::min(dtmf_index_, samples_added_per_channel);
+}
+
+void SyncBuffer::PushFrontZeros(size_t length) {
+  InsertZerosAtIndex(length, 0);
+}
+
+void SyncBuffer::InsertZerosAtIndex(size_t length, size_t position) {
+  position = std::min(position, Size());
+  length = std::min(length, Size() - position);
+  AudioMultiVector::PopBack(length);
+  for (size_t channel = 0; channel < Channels(); ++channel) {
+    channels_[channel]->InsertZerosAt(length, position);
+  }
+  if (next_index_ >= position) {
+    // We are moving the `next_index_` sample.
+    set_next_index(next_index_ + length);  // Overflow handled by subfunction.
+  }
+  if (dtmf_index_ > 0 && dtmf_index_ >= position) {
+    // We are moving the `dtmf_index_` sample.
+    set_dtmf_index(dtmf_index_ + length);  // Overflow handled by subfunction.
+  }
+}
+
+void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
+                                size_t length,
+                                size_t position) {
+  position = std::min(position, Size());  // Cap `position` in the valid range.
+  length = std::min(length, Size() - position);
+  AudioMultiVector::OverwriteAt(insert_this, length, position);
+}
+
+void SyncBuffer::ReplaceAtIndex(const AudioMultiVector& insert_this,
+                                size_t position) {
+  ReplaceAtIndex(insert_this, insert_this.Size(), position);
+}
+
+void SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
+                                         AudioFrame* output) {
+  RTC_DCHECK(output);
+  const size_t samples_to_read = std::min(FutureLength(), requested_len);
+  output->ResetWithoutMuting();
+  const size_t tot_samples_read = ReadInterleavedFromIndex(
+      next_index_, samples_to_read, output->mutable_data());
+  const size_t samples_read_per_channel = tot_samples_read / Channels();
+  next_index_ += samples_read_per_channel;
+  output->num_channels_ = Channels();
+  output->samples_per_channel_ = samples_read_per_channel;
+}
+
+void SyncBuffer::IncreaseEndTimestamp(uint32_t increment) {
+  end_timestamp_ += increment;
+}
+
+void SyncBuffer::Flush() {
+  Zeros(Size());
+  next_index_ = Size();
+  end_timestamp_ = 0;
+  dtmf_index_ = 0;
+}
+
+void SyncBuffer::set_next_index(size_t value) {
+  // Cannot set `next_index_` larger than the size of the buffer.
+  next_index_ = std::min(value, Size());
+}
+
+void SyncBuffer::set_dtmf_index(size_t value) {
+  // Cannot set `dtmf_index_` larger than the size of the buffer.
+  dtmf_index_ = std::min(value, Size());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.h b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.h
new file mode 100644
index 0000000000..cf56c432e3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer.h
@@ -0,0 +1,110 @@
+/*
+ *  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_NETEQ_SYNC_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio/audio_frame.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class SyncBuffer : public AudioMultiVector {
+ public:
+  SyncBuffer(size_t channels, size_t length)
+      : AudioMultiVector(channels, length),
+        next_index_(length),
+        end_timestamp_(0),
+        dtmf_index_(0) {}
+
+  SyncBuffer(const SyncBuffer&) = delete;
+  SyncBuffer& operator=(const SyncBuffer&) = delete;
+
+  // Returns the number of samples yet to play out from the buffer.
+  size_t FutureLength() const;
+
+  // Adds the contents of `append_this` to the back of the SyncBuffer. Removes
+  // the same number of samples from the beginning of the SyncBuffer, to
+  // maintain a constant buffer size. The `next_index_` is updated to reflect
+  // the move of the beginning of "future" data.
+  void PushBack(const AudioMultiVector& append_this) override;
+
+  // Like PushBack, but reads the samples channel-interleaved from the input.
+  void PushBackInterleaved(const rtc::BufferT<int16_t>& append_this);
+
+  // Adds `length` zeros to the beginning of each channel. Removes
+  // the same number of samples from the end of the SyncBuffer, to
+  // maintain a constant buffer size. The `next_index_` is updated to reflect
+  // the move of the beginning of "future" data.
+  // Note that this operation may delete future samples that are waiting to
+  // be played.
+  void PushFrontZeros(size_t length);
+
+  // Inserts `length` zeros into each channel at index `position`. The size of
+  // the SyncBuffer is kept constant, which means that the last `length`
+  // elements in each channel will be purged.
+  virtual void InsertZerosAtIndex(size_t length, size_t position);
+
+  // Overwrites each channel in this SyncBuffer with values taken from
+  // `insert_this`. The values are taken from the beginning of `insert_this` and
+  // are inserted starting at `position`. `length` values are written into each
+  // channel. The size of the SyncBuffer is kept constant. That is, if `length`
+  // and `position` are selected such that the new data would extend beyond the
+  // end of the current SyncBuffer, the buffer is not extended.
+  // The `next_index_` is not updated.
+  virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
+                              size_t length,
+                              size_t position);
+
+  // Same as the above method, but where all of `insert_this` is written (with
+  // the same constraints as above, that the SyncBuffer is not extended).
+  virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
+                              size_t position);
+
+  // Reads `requested_len` samples from each channel and writes them interleaved
+  // into `output`. The `next_index_` is updated to point to the sample to read
+  // next time. The AudioFrame `output` is first reset, and the `data_`,
+  // `num_channels_`, and `samples_per_channel_` fields are updated.
+  void GetNextAudioInterleaved(size_t requested_len, AudioFrame* output);
+
+  // Adds `increment` to `end_timestamp_`.
+  void IncreaseEndTimestamp(uint32_t increment);
+
+  // Flushes the buffer. The buffer will contain only zeros after the flush, and
+  // `next_index_` will point to the end, like when the buffer was first
+  // created.
+  void Flush();
+
+  const AudioVector& Channel(size_t n) const { return *channels_[n]; }
+  AudioVector& Channel(size_t n) { return *channels_[n]; }
+
+  // Accessors and mutators.
+  size_t next_index() const { return next_index_; }
+  void set_next_index(size_t value);
+  uint32_t end_timestamp() const { return end_timestamp_; }
+  void set_end_timestamp(uint32_t value) { end_timestamp_ = value; }
+  size_t dtmf_index() const { return dtmf_index_; }
+  void set_dtmf_index(size_t value);
+
+ private:
+  size_t next_index_;
+  uint32_t end_timestamp_;  // The timestamp of the last sample in the buffer.
+  size_t dtmf_index_;       // Index to the first non-DTMF sample in the buffer.
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc
new file mode 100644
index 0000000000..bdcd92446b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/sync_buffer_unittest.cc
@@ -0,0 +1,174 @@
+/*
+ *  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/neteq/sync_buffer.h"
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(SyncBuffer, CreateAndDestroy) {
+  // Create a SyncBuffer with two channels and 10 samples each.
+  static const size_t kLen = 10;
+  static const size_t kChannels = 2;
+  SyncBuffer sync_buffer(kChannels, kLen);
+  EXPECT_EQ(kChannels, sync_buffer.Channels());
+  EXPECT_EQ(kLen, sync_buffer.Size());
+  // When the buffer is empty, the next index to play out is at the end.
+  EXPECT_EQ(kLen, sync_buffer.next_index());
+  // Verify that all elements are zero.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kLen; ++i) {
+      EXPECT_EQ(0, sync_buffer[channel][i]);
+    }
+  }
+}
+
+TEST(SyncBuffer, SetNextIndex) {
+  // Create a SyncBuffer with two channels and 100 samples each.
+  static const size_t kLen = 100;
+  static const size_t kChannels = 2;
+  SyncBuffer sync_buffer(kChannels, kLen);
+  sync_buffer.set_next_index(0);
+  EXPECT_EQ(0u, sync_buffer.next_index());
+  sync_buffer.set_next_index(kLen / 2);
+  EXPECT_EQ(kLen / 2, sync_buffer.next_index());
+  sync_buffer.set_next_index(kLen);
+  EXPECT_EQ(kLen, sync_buffer.next_index());
+  // Try to set larger than the buffer size; should cap at buffer size.
+  sync_buffer.set_next_index(kLen + 1);
+  EXPECT_EQ(kLen, sync_buffer.next_index());
+}
+
+TEST(SyncBuffer, PushBackAndFlush) {
+  // Create a SyncBuffer with two channels and 100 samples each.
+  static const size_t kLen = 100;
+  static const size_t kChannels = 2;
+  SyncBuffer sync_buffer(kChannels, kLen);
+  static const size_t kNewLen = 10;
+  AudioMultiVector new_data(kChannels, kNewLen);
+  // Populate `new_data`.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kNewLen; ++i) {
+      new_data[channel][i] = rtc::checked_cast<int16_t>(i);
+    }
+  }
+  // Push back `new_data` into `sync_buffer`. This operation should pop out
+  // data from the front of `sync_buffer`, so that the size of the buffer
+  // remains the same. The `next_index_` should also move with the same length.
+  sync_buffer.PushBack(new_data);
+  ASSERT_EQ(kLen, sync_buffer.Size());
+  // Verify that `next_index_` moved accordingly.
+  EXPECT_EQ(kLen - kNewLen, sync_buffer.next_index());
+  // Verify the new contents.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kNewLen; ++i) {
+      EXPECT_EQ(new_data[channel][i],
+                sync_buffer[channel][sync_buffer.next_index() + i]);
+    }
+  }
+
+  // Now flush the buffer, and verify that it is all zeros, and that next_index
+  // points to the end.
+  sync_buffer.Flush();
+  ASSERT_EQ(kLen, sync_buffer.Size());
+  EXPECT_EQ(kLen, sync_buffer.next_index());
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kLen; ++i) {
+      EXPECT_EQ(0, sync_buffer[channel][i]);
+    }
+  }
+}
+
+TEST(SyncBuffer, PushFrontZeros) {
+  // Create a SyncBuffer with two channels and 100 samples each.
+  static const size_t kLen = 100;
+  static const size_t kChannels = 2;
+  SyncBuffer sync_buffer(kChannels, kLen);
+  static const size_t kNewLen = 10;
+  AudioMultiVector new_data(kChannels, kNewLen);
+  // Populate `new_data`.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kNewLen; ++i) {
+      new_data[channel][i] = rtc::checked_cast<int16_t>(1000 + i);
+    }
+  }
+  sync_buffer.PushBack(new_data);
+  EXPECT_EQ(kLen, sync_buffer.Size());
+
+  // Push `kNewLen` - 1 zeros into each channel in the front of the SyncBuffer.
+  sync_buffer.PushFrontZeros(kNewLen - 1);
+  EXPECT_EQ(kLen, sync_buffer.Size());  // Size should remain the same.
+  // Verify that `next_index_` moved accordingly. Should be at the end - 1.
+  EXPECT_EQ(kLen - 1, sync_buffer.next_index());
+  // Verify the zeros.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kNewLen - 1; ++i) {
+      EXPECT_EQ(0, sync_buffer[channel][i]);
+    }
+  }
+  // Verify that the correct data is at the end of the SyncBuffer.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    EXPECT_EQ(1000, sync_buffer[channel][sync_buffer.next_index()]);
+  }
+}
+
+TEST(SyncBuffer, GetNextAudioInterleaved) {
+  // Create a SyncBuffer with two channels and 100 samples each.
+  static const size_t kLen = 100;
+  static const size_t kChannels = 2;
+  SyncBuffer sync_buffer(kChannels, kLen);
+  static const size_t kNewLen = 10;
+  AudioMultiVector new_data(kChannels, kNewLen);
+  // Populate `new_data`.
+  for (size_t channel = 0; channel < kChannels; ++channel) {
+    for (size_t i = 0; i < kNewLen; ++i) {
+      new_data[channel][i] = rtc::checked_cast<int16_t>(i);
+    }
+  }
+  // Push back `new_data` into `sync_buffer`. This operation should pop out
+  // data from the front of `sync_buffer`, so that the size of the buffer
+  // remains the same. The `next_index_` should also move with the same length.
+  sync_buffer.PushBack(new_data);
+
+  // Read to interleaved output. Read in two batches, where each read operation
+  // should automatically update the `net_index_` in the SyncBuffer.
+  // Note that `samples_read` is the number of samples read from each channel.
+  // That is, the number of samples written to `output` is
+  // `samples_read` * `kChannels`.
+  AudioFrame output1;
+  sync_buffer.GetNextAudioInterleaved(kNewLen / 2, &output1);
+  EXPECT_EQ(kChannels, output1.num_channels_);
+  EXPECT_EQ(kNewLen / 2, output1.samples_per_channel_);
+
+  AudioFrame output2;
+  sync_buffer.GetNextAudioInterleaved(kNewLen / 2, &output2);
+  EXPECT_EQ(kChannels, output2.num_channels_);
+  EXPECT_EQ(kNewLen / 2, output2.samples_per_channel_);
+
+  // Verify the data.
+  const int16_t* output_ptr = output1.data();
+  for (size_t i = 0; i < kNewLen / 2; ++i) {
+    for (size_t channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(new_data[channel][i], *output_ptr);
+      ++output_ptr;
+    }
+  }
+  output_ptr = output2.data();
+  for (size_t i = kNewLen / 2; i < kNewLen; ++i) {
+    for (size_t channel = 0; channel < kChannels; ++channel) {
+      EXPECT_EQ(new_data[channel][i], *output_ptr);
+      ++output_ptr;
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m
new file mode 100644
index 0000000000..031d8a39ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m
@@ -0,0 +1,201 @@
+%
+%  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.
+%
+
+function outStruct = parse_delay_file(file)
+
+fid = fopen(file, 'rb');
+if fid == -1
+    error('Cannot open file %s', file);
+end
+
+textline = fgetl(fid);
+if ~strncmp(textline, '#!NetEQ_Delay_Logging', 21)
+    error('Wrong file format');
+end
+
+ver = sscanf(textline, '#!NetEQ_Delay_Logging%d.%d');
+if ~all(ver == [2; 0])
+    error('Wrong version of delay logging function')
+end
+
+
+start_pos = ftell(fid);
+fseek(fid, -12, 'eof');
+textline = fgetl(fid);
+if ~strncmp(textline, 'End of file', 21)
+    error('File ending is not correct. Seems like the simulation ended abnormally.');
+end
+
+fseek(fid,-12-4, 'eof');
+Npackets = fread(fid, 1, 'int32');
+fseek(fid, start_pos, 'bof');
+
+rtpts = zeros(Npackets, 1);
+seqno = zeros(Npackets, 1);
+pt = zeros(Npackets, 1);
+plen = zeros(Npackets, 1);
+recin_t = nan*ones(Npackets, 1);
+decode_t = nan*ones(Npackets, 1);
+playout_delay = zeros(Npackets, 1);
+optbuf = zeros(Npackets, 1);
+
+fs_ix = 1;
+clock = 0;
+ts_ix = 1;
+ended = 0;
+late_packets = 0;
+fs_now = 8000;
+last_decode_k = 0;
+tot_expand = 0;
+tot_accelerate = 0;
+tot_preemptive = 0;
+
+while not(ended)
+    signal = fread(fid, 1, '*int32');
+    
+    switch signal
+        case 3 % NETEQ_DELAY_LOGGING_SIGNAL_CLOCK
+            clock = fread(fid, 1, '*float32');
+            
+            % keep on reading batches of M until the signal is no longer "3"
+            % read int32 + float32 in one go
+            % this is to save execution time
+            temp = [3; 0];
+            M = 120;
+            while all(temp(1,:) == 3)
+                fp = ftell(fid);
+                temp = fread(fid, [2 M], '*int32');
+            end
+            
+            % back up to last clock event
+            fseek(fid, fp - ftell(fid) + ...
+                (find(temp(1,:) ~= 3, 1 ) - 2) * 2 * 4 + 4, 'cof');
+            % read the last clock value
+            clock = fread(fid, 1, '*float32');
+            
+        case 1 % NETEQ_DELAY_LOGGING_SIGNAL_RECIN
+            temp_ts = fread(fid, 1, 'uint32');
+            
+            if late_packets > 0
+                temp_ix = ts_ix - 1;
+                while (temp_ix >= 1) && (rtpts(temp_ix) ~= temp_ts)
+                    % TODO(hlundin): use matlab vector search instead?
+                    temp_ix = temp_ix - 1;
+                end
+                
+                if temp_ix >= 1
+                    % the ts was found in the vector
+                    late_packets = late_packets - 1;
+                else
+                    temp_ix = ts_ix;
+                    ts_ix = ts_ix + 1;
+                end
+            else
+                temp_ix = ts_ix;
+                ts_ix = ts_ix + 1;
+            end
+            
+            rtpts(temp_ix) = temp_ts;
+            seqno(temp_ix) = fread(fid, 1, 'uint16');
+            pt(temp_ix) = fread(fid, 1, 'int32');
+            plen(temp_ix) = fread(fid, 1, 'int16');
+            recin_t(temp_ix) = clock;
+            
+        case 2 % NETEQ_DELAY_LOGGING_SIGNAL_FLUSH
+            % do nothing
+            
+        case 4 % NETEQ_DELAY_LOGGING_SIGNAL_EOF
+            ended = 1;
+            
+        case 5 % NETEQ_DELAY_LOGGING_SIGNAL_DECODE
+            last_decode_ts = fread(fid, 1, 'uint32');
+            temp_delay = fread(fid, 1, 'uint16');
+            
+            k = find(rtpts(1:(ts_ix - 1))==last_decode_ts,1,'last');
+            if ~isempty(k)
+                decode_t(k) = clock;
+                playout_delay(k) = temp_delay + ...
+                    5 *  fs_now / 8000; % add overlap length
+                last_decode_k = k;
+            end
+            
+        case 6 % NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS
+            fsvec(fs_ix) = fread(fid, 1, 'uint16');
+            fschange_ts(fs_ix) = last_decode_ts;
+            fs_now = fsvec(fs_ix);
+            fs_ix = fs_ix + 1;
+            
+        case 7 % NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO
+            playout_delay(last_decode_k) = playout_delay(last_decode_k) ...
+                + fread(fid, 1, 'int32');
+            
+        case 8 % NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO
+            temp = fread(fid, 1, 'int32');
+            if last_decode_k ~= 0
+                tot_expand = tot_expand + temp / (fs_now / 1000);
+            end                
+            
+        case 9 % NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO
+            temp = fread(fid, 1, 'int32');
+            if last_decode_k ~= 0
+                tot_accelerate = tot_accelerate + temp / (fs_now / 1000);
+            end                
+
+        case 10 % NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO
+            temp = fread(fid, 1, 'int32');
+            if last_decode_k ~= 0
+                tot_preemptive = tot_preemptive + temp / (fs_now / 1000);
+            end                
+            
+        case 11 % NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF
+            optbuf(last_decode_k) = fread(fid, 1, 'int32');
+            
+        case 12 % NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC
+            last_decode_ts = fread(fid, 1, 'uint32');
+            k = ts_ix - 1;
+            
+            while (k >= 1) && (rtpts(k) ~= last_decode_ts)
+                % TODO(hlundin): use matlab vector search instead?
+                k = k - 1;
+            end
+            
+            if k < 1
+                % packet not received yet
+                k = ts_ix;
+                rtpts(ts_ix) = last_decode_ts;
+                late_packets = late_packets + 1;
+            end
+            
+            decode_t(k) = clock;
+            playout_delay(k) = fread(fid, 1, 'uint16') + ...
+                5 *  fs_now / 8000; % add overlap length
+            last_decode_k = k;
+             
+    end
+    
+end
+
+
+fclose(fid);
+
+outStruct = struct(...
+    'ts', rtpts, ...
+    'sn', seqno, ...
+    'pt', pt,...
+    'plen', plen,...
+    'arrival', recin_t,...
+    'decode', decode_t,...
+    'fs', fsvec(:),...
+    'fschange_ts', fschange_ts(:),...
+    'playout_delay', playout_delay,...
+    'tot_expand', tot_expand,...
+    'tot_accelerate', tot_accelerate,...
+    'tot_preemptive', tot_preemptive,...
+    'optbuf', optbuf);
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m
new file mode 100644
index 0000000000..86d533fbeb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m
@@ -0,0 +1,197 @@
+%
+%  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.
+%
+
+function [delay_struct, delayvalues] = plot_neteq_delay(delayfile, varargin)
+
+% InfoStruct = plot_neteq_delay(delayfile)
+% InfoStruct = plot_neteq_delay(delayfile, 'skipdelay', skip_seconds)
+%
+% Henrik Lundin, 2006-11-17
+% Henrik Lundin, 2011-05-17
+%
+
+try
+    s = parse_delay_file(delayfile);
+catch
+    error(lasterr);
+end
+
+delayskip=0;
+noplot=0;
+arg_ptr=1;
+delaypoints=[];
+
+s.sn=unwrap_seqno(s.sn);
+
+while arg_ptr+1 <= nargin
+    switch lower(varargin{arg_ptr})
+    case {'skipdelay', 'delayskip'}
+        % skip a number of seconds in the beginning when calculating delays
+        delayskip = varargin{arg_ptr+1};
+        arg_ptr = arg_ptr + 2;
+    case 'noplot'
+        noplot=1;
+        arg_ptr = arg_ptr + 1;
+    case {'get_delay', 'getdelay'}
+        % return a vector of delay values for the points in the given vector
+        delaypoints = varargin{arg_ptr+1};
+        arg_ptr = arg_ptr + 2;
+    otherwise
+        warning('Unknown switch %s\n', varargin{arg_ptr});
+        arg_ptr = arg_ptr + 1;
+    end
+end
+
+% find lost frames that were covered by one-descriptor decoding
+one_desc_ix=find(isnan(s.arrival));
+for k=1:length(one_desc_ix)
+    ix=find(s.ts==max(s.ts(s.ts(one_desc_ix(k))>s.ts)));
+    s.sn(one_desc_ix(k))=s.sn(ix)+1;
+    s.pt(one_desc_ix(k))=s.pt(ix);
+    s.arrival(one_desc_ix(k))=s.arrival(ix)+s.decode(one_desc_ix(k))-s.decode(ix);
+end
+
+% remove duplicate received frames that were never decoded (RED codec)
+if length(unique(s.ts(isfinite(s.ts)))) < length(s.ts(isfinite(s.ts)))
+    ix=find(isfinite(s.decode));
+    s.sn=s.sn(ix);
+    s.ts=s.ts(ix);
+    s.arrival=s.arrival(ix);
+    s.playout_delay=s.playout_delay(ix);
+    s.pt=s.pt(ix);
+    s.optbuf=s.optbuf(ix);
+    plen=plen(ix);
+    s.decode=s.decode(ix);
+end
+
+% find non-unique sequence numbers
+[~,un_ix]=unique(s.sn);
+nonun_ix=setdiff(1:length(s.sn),un_ix);
+if ~isempty(nonun_ix)
+    warning('RTP sequence numbers are in error');
+end
+            
+% sort vectors
+[s.sn,sort_ix]=sort(s.sn);
+s.ts=s.ts(sort_ix);
+s.arrival=s.arrival(sort_ix);
+s.decode=s.decode(sort_ix);
+s.playout_delay=s.playout_delay(sort_ix);
+s.pt=s.pt(sort_ix);
+
+send_t=s.ts-s.ts(1);
+if length(s.fs)<1
+    warning('No info about sample rate found in file. Using default 8000.');
+    s.fs(1)=8000;
+    s.fschange_ts(1)=min(s.ts);
+elseif s.fschange_ts(1)>min(s.ts)
+    s.fschange_ts(1)=min(s.ts);
+end
+
+end_ix=length(send_t);
+for k=length(s.fs):-1:1
+    start_ix=find(s.ts==s.fschange_ts(k));
+    send_t(start_ix:end_ix)=send_t(start_ix:end_ix)/s.fs(k)*1000;
+    s.playout_delay(start_ix:end_ix)=s.playout_delay(start_ix:end_ix)/s.fs(k)*1000;
+    s.optbuf(start_ix:end_ix)=s.optbuf(start_ix:end_ix)/s.fs(k)*1000;
+    end_ix=start_ix-1;
+end
+
+tot_time=max(send_t)-min(send_t);
+
+seq_ix=s.sn-min(s.sn)+1;
+send_t=send_t+max(min(s.arrival-send_t),0);
+
+plot_send_t=nan*ones(max(seq_ix),1);
+plot_send_t(seq_ix)=send_t;
+plot_nw_delay=nan*ones(max(seq_ix),1);
+plot_nw_delay(seq_ix)=s.arrival-send_t;
+
+cng_ix=find(s.pt~=13); % find those packets that are not CNG/SID
+    
+if noplot==0
+    h=plot(plot_send_t/1000,plot_nw_delay);
+    set(h,'color',0.75*[1 1 1]);
+    hold on
+    if any(s.optbuf~=0)
+        peak_ix=find(s.optbuf(cng_ix)<0); % peak mode is labeled with negative values
+        no_peak_ix=find(s.optbuf(cng_ix)>0); %setdiff(1:length(cng_ix),peak_ix);
+        h1=plot(send_t(cng_ix(peak_ix))/1000,...
+            s.arrival(cng_ix(peak_ix))+abs(s.optbuf(cng_ix(peak_ix)))-send_t(cng_ix(peak_ix)),...
+            'r.');
+        h2=plot(send_t(cng_ix(no_peak_ix))/1000,...
+            s.arrival(cng_ix(no_peak_ix))+abs(s.optbuf(cng_ix(no_peak_ix)))-send_t(cng_ix(no_peak_ix)),...
+            'g.');
+        set([h1, h2],'markersize',1)
+    end
+    %h=plot(send_t(seq_ix)/1000,s.decode+s.playout_delay-send_t(seq_ix));
+    h=plot(send_t(cng_ix)/1000,s.decode(cng_ix)+s.playout_delay(cng_ix)-send_t(cng_ix));
+    set(h,'linew',1.5);
+    hold off
+    ax1=axis;
+    axis tight
+    ax2=axis;
+    axis([ax2(1:3) ax1(4)])
+end
+
+
+% calculate delays and other parameters
+
+delayskip_ix = find(send_t-send_t(1)>=delayskip*1000, 1 );
+
+use_ix = intersect(cng_ix,... % use those that are not CNG/SID frames...
+    intersect(find(isfinite(s.decode)),... % ... that did arrive ...
+    (delayskip_ix:length(s.decode))')); % ... and are sent after delayskip seconds
+
+mean_delay = mean(s.decode(use_ix)+s.playout_delay(use_ix)-send_t(use_ix));
+neteq_delay = mean(s.decode(use_ix)+s.playout_delay(use_ix)-s.arrival(use_ix));
+
+Npack=max(s.sn(delayskip_ix:end))-min(s.sn(delayskip_ix:end))+1;
+nw_lossrate=(Npack-length(s.sn(delayskip_ix:end)))/Npack;
+neteq_lossrate=(length(s.sn(delayskip_ix:end))-length(use_ix))/Npack;
+
+delay_struct=struct('mean_delay',mean_delay,'neteq_delay',neteq_delay,...
+    'nw_lossrate',nw_lossrate,'neteq_lossrate',neteq_lossrate,...
+    'tot_expand',round(s.tot_expand),'tot_accelerate',round(s.tot_accelerate),...
+    'tot_preemptive',round(s.tot_preemptive),'tot_time',tot_time,...
+    'filename',delayfile,'units','ms','fs',unique(s.fs));
+    
+if not(isempty(delaypoints))
+    delayvalues=interp1(send_t(cng_ix),...
+        s.decode(cng_ix)+s.playout_delay(cng_ix)-send_t(cng_ix),...
+        delaypoints,'nearest',NaN);
+else
+    delayvalues=[];
+end
+
+
+
+% SUBFUNCTIONS %
+
+function y=unwrap_seqno(x)
+
+jumps=find(abs((diff(x)-1))>65000);
+
+while ~isempty(jumps)
+    n=jumps(1);
+    if x(n+1)-x(n) < 0
+        % negative jump
+        x(n+1:end)=x(n+1:end)+65536;
+    else
+        % positive jump
+        x(n+1:end)=x(n+1:end)-65536;
+    end
+    
+    jumps=find(abs((diff(x(n+1:end))-1))>65000);
+end
+
+y=x;
+
+return;
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc
new file mode 100644
index 0000000000..e6c1809fb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc
@@ -0,0 +1,423 @@
+/*
+ *  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/neteq/test/neteq_decoding_test.h"
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/rtp_headers.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/test/result_sink.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/testsupport/file_utils.h"
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#else
+#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+namespace {
+
+void LoadDecoders(webrtc::NetEq* neteq) {
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(0, SdpAudioFormat("pcmu", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(8, SdpAudioFormat("pcma", 8000, 1)));
+#ifdef WEBRTC_CODEC_ILBC
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(102, SdpAudioFormat("ilbc", 8000, 1)));
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(103, SdpAudioFormat("isac", 16000, 1)));
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(104, SdpAudioFormat("isac", 32000, 1)));
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(
+                111, SdpAudioFormat("opus", 48000, 2, {{"stereo", "0"}})));
+#endif
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(93, SdpAudioFormat("L16", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(94, SdpAudioFormat("L16", 16000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(95, SdpAudioFormat("L16", 32000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(13, SdpAudioFormat("cn", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(98, SdpAudioFormat("cn", 16000, 1)));
+}
+
+}  // namespace
+
+const int NetEqDecodingTest::kTimeStepMs;
+const size_t NetEqDecodingTest::kBlockSize8kHz;
+const size_t NetEqDecodingTest::kBlockSize16kHz;
+const size_t NetEqDecodingTest::kBlockSize32kHz;
+const int NetEqDecodingTest::kInitSampleRateHz;
+
+NetEqDecodingTest::NetEqDecodingTest()
+    : clock_(0),
+      config_(),
+      output_sample_rate_(kInitSampleRateHz),
+      algorithmic_delay_ms_(0) {
+  config_.sample_rate_hz = kInitSampleRateHz;
+}
+
+void NetEqDecodingTest::SetUp() {
+  auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+  neteq_ = DefaultNetEqFactory().CreateNetEq(config_, decoder_factory, &clock_);
+  NetEqNetworkStatistics stat;
+  ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
+  algorithmic_delay_ms_ = stat.current_buffer_size_ms;
+  ASSERT_TRUE(neteq_);
+  LoadDecoders(neteq_.get());
+}
+
+void NetEqDecodingTest::TearDown() {}
+
+void NetEqDecodingTest::OpenInputFile(absl::string_view rtp_file) {
+  rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
+}
+
+void NetEqDecodingTest::Process() {
+  // Check if time to receive.
+  while (packet_ && clock_.TimeInMilliseconds() >= packet_->time_ms()) {
+    if (packet_->payload_length_bytes() > 0) {
+#ifndef WEBRTC_CODEC_ISAC
+      // Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
+      if (packet_->header().payloadType != 104)
+#endif
+        ASSERT_EQ(
+            0, neteq_->InsertPacket(
+                   packet_->header(),
+                   rtc::ArrayView<const uint8_t>(
+                       packet_->payload(), packet_->payload_length_bytes())));
+    }
+    // Get next packet.
+    packet_ = rtp_source_->NextPacket();
+  }
+
+  // Get audio from NetEq.
+  bool muted;
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
+  ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize16kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize32kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize48kHz));
+  output_sample_rate_ = out_frame_.sample_rate_hz_;
+  EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
+
+  // Increase time.
+  clock_.AdvanceTimeMilliseconds(kTimeStepMs);
+}
+
+void NetEqDecodingTest::DecodeAndCompare(
+    absl::string_view rtp_file,
+    absl::string_view output_checksum,
+    absl::string_view network_stats_checksum,
+    bool gen_ref) {
+  OpenInputFile(rtp_file);
+
+  std::string ref_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_universal_ref.pcm" : "";
+  ResultSink output(ref_out_file);
+
+  std::string stat_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : "";
+  ResultSink network_stats(stat_out_file);
+
+  packet_ = rtp_source_->NextPacket();
+  int i = 0;
+  uint64_t last_concealed_samples = 0;
+  uint64_t last_total_samples_received = 0;
+  while (packet_) {
+    rtc::StringBuilder ss;
+    ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
+    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+    ASSERT_NO_FATAL_FAILURE(Process());
+    ASSERT_NO_FATAL_FAILURE(
+        output.AddResult(out_frame_.data(), out_frame_.samples_per_channel_));
+
+    // Query the network statistics API once per second
+    if (clock_.TimeInMilliseconds() % 1000 == 0) {
+      // Process NetworkStatistics.
+      NetEqNetworkStatistics current_network_stats;
+      ASSERT_EQ(0, neteq_->NetworkStatistics(&current_network_stats));
+      ASSERT_NO_FATAL_FAILURE(network_stats.AddResult(current_network_stats));
+
+      // Verify that liftime stats and network stats report similar loss
+      // concealment rates.
+      auto lifetime_stats = neteq_->GetLifetimeStatistics();
+      const uint64_t delta_concealed_samples =
+          lifetime_stats.concealed_samples - last_concealed_samples;
+      last_concealed_samples = lifetime_stats.concealed_samples;
+      const uint64_t delta_total_samples_received =
+          lifetime_stats.total_samples_received - last_total_samples_received;
+      last_total_samples_received = lifetime_stats.total_samples_received;
+      // The tolerance is 1% but expressed in Q14.
+      EXPECT_NEAR(
+          (delta_concealed_samples << 14) / delta_total_samples_received,
+          current_network_stats.expand_rate, (2 << 14) / 100.0);
+    }
+  }
+
+  SCOPED_TRACE("Check output audio.");
+  output.VerifyChecksum(output_checksum);
+  SCOPED_TRACE("Check network stats.");
+  network_stats.VerifyChecksum(network_stats_checksum);
+}
+
+void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
+                                        int timestamp,
+                                        RTPHeader* rtp_info) {
+  rtp_info->sequenceNumber = frame_index;
+  rtp_info->timestamp = timestamp;
+  rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info->payloadType = 94;  // PCM16b WB codec.
+  rtp_info->markerBit = false;
+}
+
+void NetEqDecodingTest::PopulateCng(int frame_index,
+                                    int timestamp,
+                                    RTPHeader* rtp_info,
+                                    uint8_t* payload,
+                                    size_t* payload_len) {
+  rtp_info->sequenceNumber = frame_index;
+  rtp_info->timestamp = timestamp;
+  rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info->payloadType = 98;  // WB CNG.
+  rtp_info->markerBit = false;
+  payload[0] = 64;   // Noise level -64 dBov, quite arbitrarily chosen.
+  *payload_len = 1;  // Only noise level, no spectral parameters.
+}
+
+void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
+                                 uint32_t start_timestamp,
+                                 const std::set<uint16_t>& drop_seq_numbers,
+                                 bool expect_seq_no_wrap,
+                                 bool expect_timestamp_wrap) {
+  uint16_t seq_no = start_seq_no;
+  uint32_t timestamp = start_timestamp;
+  const int kBlocksPerFrame = 3;  // Number of 10 ms blocks per frame.
+  const int kFrameSizeMs = kBlocksPerFrame * kTimeStepMs;
+  const int kSamples = kBlockSize16kHz * kBlocksPerFrame;
+  const size_t kPayloadBytes = kSamples * sizeof(int16_t);
+  double next_input_time_ms = 0.0;
+
+  // Insert speech for 2 seconds.
+  const int kSpeechDurationMs = 2000;
+  uint16_t last_seq_no;
+  uint32_t last_timestamp;
+  bool timestamp_wrapped = false;
+  bool seq_no_wrapped = false;
+  for (double t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) {
+        // This sequence number was not in the set to drop. Insert it.
+        ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      }
+      NetEqNetworkStatistics network_stats;
+      ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
+
+      EXPECT_LE(network_stats.preferred_buffer_size_ms, 80);
+      EXPECT_LE(network_stats.current_buffer_size_ms,
+                80 + algorithmic_delay_ms_);
+      last_seq_no = seq_no;
+      last_timestamp = timestamp;
+
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += static_cast<double>(kFrameSizeMs);
+
+      seq_no_wrapped |= seq_no < last_seq_no;
+      timestamp_wrapped |= timestamp < last_timestamp;
+    }
+    // Pull out data once.
+    AudioFrame output;
+    bool muted;
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+    ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
+    ASSERT_EQ(1u, output.num_channels_);
+
+    // Expect delay (in samples) to be less than 2 packets.
+    absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+    ASSERT_TRUE(playout_timestamp);
+    EXPECT_LE(timestamp - *playout_timestamp,
+              static_cast<uint32_t>(kSamples * 2));
+  }
+  // Make sure we have actually tested wrap-around.
+  ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
+  ASSERT_EQ(expect_timestamp_wrap, timestamp_wrapped);
+}
+
+void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
+                                              double network_freeze_ms,
+                                              bool pull_audio_during_freeze,
+                                              int delay_tolerance_ms,
+                                              int max_time_to_speech_ms) {
+  uint16_t seq_no = 0;
+  uint32_t timestamp = 0;
+  const int kFrameSizeMs = 30;
+  const size_t kSamples = kFrameSizeMs * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  double next_input_time_ms = 0.0;
+  double t_ms;
+  bool muted;
+
+  // Insert speech for 5 seconds.
+  const int kSpeechDurationMs = 5000;
+  for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+  absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+  ASSERT_TRUE(playout_timestamp);
+  int32_t delay_before = timestamp - *playout_timestamp;
+
+  // Insert CNG for 1 minute (= 60000 ms).
+  const int kCngPeriodMs = 100;
+  const int kCngPeriodSamples = kCngPeriodMs * 16;  // Period in 16 kHz samples.
+  const int kCngDurationMs = 60000;
+  for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one CNG frame each 100 ms.
+      uint8_t payload[kPayloadBytes];
+      size_t payload_len;
+      RTPHeader rtp_info;
+      PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, payload_len)));
+      ++seq_no;
+      timestamp += kCngPeriodSamples;
+      next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+
+  if (network_freeze_ms > 0) {
+    // First keep pulling audio for `network_freeze_ms` without inserting
+    // any data, then insert CNG data corresponding to `network_freeze_ms`
+    // without pulling any output audio.
+    const double loop_end_time = t_ms + network_freeze_ms;
+    for (; t_ms < loop_end_time; t_ms += 10) {
+      // Pull out data once.
+      ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+      ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+      EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+    }
+    bool pull_once = pull_audio_during_freeze;
+    // If `pull_once` is true, GetAudio will be called once half-way through
+    // the network recovery period.
+    double pull_time_ms = (t_ms + next_input_time_ms) / 2;
+    while (next_input_time_ms <= t_ms) {
+      if (pull_once && next_input_time_ms >= pull_time_ms) {
+        pull_once = false;
+        // Pull out data once.
+        ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+        ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+        EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+        t_ms += 10;
+      }
+      // Insert one CNG frame each 100 ms.
+      uint8_t payload[kPayloadBytes];
+      size_t payload_len;
+      RTPHeader rtp_info;
+      PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, payload_len)));
+      ++seq_no;
+      timestamp += kCngPeriodSamples;
+      next_input_time_ms += kCngPeriodMs * drift_factor;
+    }
+  }
+
+  // Insert speech again until output type is speech.
+  double speech_restart_time_ms = t_ms;
+  while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += kFrameSizeMs * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+    // Increase clock.
+    t_ms += 10;
+  }
+
+  // Check that the speech starts again within reasonable time.
+  double time_until_speech_returns_ms = t_ms - speech_restart_time_ms;
+  EXPECT_LT(time_until_speech_returns_ms, max_time_to_speech_ms);
+  playout_timestamp = neteq_->GetPlayoutTimestamp();
+  ASSERT_TRUE(playout_timestamp);
+  int32_t delay_after = timestamp - *playout_timestamp;
+  // Compare delay before and after, and make sure it differs less than 20 ms.
+  EXPECT_LE(delay_after, delay_before + delay_tolerance_ms * 16);
+  EXPECT_GE(delay_after, delay_before - delay_tolerance_ms * 16);
+}
+
+void NetEqDecodingTestTwoInstances::SetUp() {
+  NetEqDecodingTest::SetUp();
+  config2_ = config_;
+}
+
+void NetEqDecodingTestTwoInstances::CreateSecondInstance() {
+  auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+  neteq2_ =
+      DefaultNetEqFactory().CreateNetEq(config2_, decoder_factory, &clock_);
+  ASSERT_TRUE(neteq2_);
+  LoadDecoders(neteq2_.get());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h
new file mode 100644
index 0000000000..456c397fdd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h
@@ -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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
+
+#include <memory>
+#include <set>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/audio/audio_frame.h"
+#include "api/neteq/neteq.h"
+#include "api/rtp_headers.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class NetEqDecodingTest : public ::testing::Test {
+ protected:
+  // NetEQ must be polled for data once every 10 ms.
+  // Thus, none of the constants below can be changed.
+  static constexpr int kTimeStepMs = 10;
+  static constexpr size_t kBlockSize8kHz = kTimeStepMs * 8;
+  static constexpr size_t kBlockSize16kHz = kTimeStepMs * 16;
+  static constexpr size_t kBlockSize32kHz = kTimeStepMs * 32;
+  static constexpr size_t kBlockSize48kHz = kTimeStepMs * 48;
+  static constexpr int kInitSampleRateHz = 8000;
+
+  NetEqDecodingTest();
+  virtual void SetUp();
+  virtual void TearDown();
+  void OpenInputFile(absl::string_view rtp_file);
+  void Process();
+
+  void DecodeAndCompare(absl::string_view rtp_file,
+                        absl::string_view output_checksum,
+                        absl::string_view network_stats_checksum,
+                        bool gen_ref);
+
+  static void PopulateRtpInfo(int frame_index,
+                              int timestamp,
+                              RTPHeader* rtp_info);
+  static void PopulateCng(int frame_index,
+                          int timestamp,
+                          RTPHeader* rtp_info,
+                          uint8_t* payload,
+                          size_t* payload_len);
+
+  void WrapTest(uint16_t start_seq_no,
+                uint32_t start_timestamp,
+                const std::set<uint16_t>& drop_seq_numbers,
+                bool expect_seq_no_wrap,
+                bool expect_timestamp_wrap);
+
+  void LongCngWithClockDrift(double drift_factor,
+                             double network_freeze_ms,
+                             bool pull_audio_during_freeze,
+                             int delay_tolerance_ms,
+                             int max_time_to_speech_ms);
+
+  SimulatedClock clock_;
+  std::unique_ptr<NetEq> neteq_;
+  NetEq::Config config_;
+  std::unique_ptr<test::RtpFileSource> rtp_source_;
+  std::unique_ptr<test::Packet> packet_;
+  AudioFrame out_frame_;
+  int output_sample_rate_;
+  int algorithmic_delay_ms_;
+};
+
+class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
+ public:
+  NetEqDecodingTestTwoInstances() : NetEqDecodingTest() {}
+
+  void SetUp() override;
+
+  void CreateSecondInstance();
+
+ protected:
+  std::unique_ptr<NetEq> neteq2_;
+  NetEq::Config config2_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
new file mode 100644
index 0000000000..1004141f16
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
@@ -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.
+ */
+
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 8;
+static const int kOutputSampleRateKhz = 8;
+}  // namespace
+
+class NetEqIlbcQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqIlbcQualityTest()
+      : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         SdpAudioFormat("ilbc", 8000, 1)) {
+    // Flag validation
+    RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) == 20 ||
+              absl::GetFlag(FLAGS_frame_size_ms) == 30 ||
+              absl::GetFlag(FLAGS_frame_size_ms) == 40 ||
+              absl::GetFlag(FLAGS_frame_size_ms) == 60)
+        << "Invalid frame size, should be 20, 30, 40, or 60 ms.";
+  }
+
+  void SetUp() override {
+    ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio.";
+    AudioEncoderIlbcConfig config;
+    config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+    encoder_.reset(new AudioEncoderIlbcImpl(config, 102));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  std::unique_ptr<AudioEncoderIlbcImpl> encoder_;
+};
+
+TEST_F(NetEqIlbcQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
new file mode 100644
index 0000000000..5a2df24ef6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -0,0 +1,183 @@
+/*
+ *  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 "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+
+ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps).");
+
+ABSL_FLAG(int,
+          complexity,
+          10,
+          "Complexity: 0 ~ 10 -- defined as in Opus"
+          "specification.");
+
+ABSL_FLAG(int, maxplaybackrate, 48000, "Maximum playback rate (Hz).");
+
+ABSL_FLAG(int, application, 0, "Application mode: 0 -- VOIP, 1 -- Audio.");
+
+ABSL_FLAG(int, reported_loss_rate, 10, "Reported percentile of packet loss.");
+
+ABSL_FLAG(bool, fec, false, "Enable FEC for encoding (-nofec to disable).");
+
+ABSL_FLAG(bool, dtx, false, "Enable DTX for encoding (-nodtx to disable).");
+
+ABSL_FLAG(int, sub_packets, 1, "Number of sub packets to repacketize.");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+
+static const int kOpusBlockDurationMs = 20;
+static const int kOpusSamplingKhz = 48;
+}  // namespace
+
+class NetEqOpusQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqOpusQualityTest();
+  void SetUp() override;
+  void TearDown() override;
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override;
+
+ private:
+  WebRtcOpusEncInst* opus_encoder_;
+  OpusRepacketizer* repacketizer_;
+  size_t sub_block_size_samples_;
+  int bit_rate_kbps_;
+  bool fec_;
+  bool dtx_;
+  int complexity_;
+  int maxplaybackrate_;
+  int target_loss_rate_;
+  int sub_packets_;
+  int application_;
+};
+
+NetEqOpusQualityTest::NetEqOpusQualityTest()
+    : NetEqQualityTest(kOpusBlockDurationMs * absl::GetFlag(FLAGS_sub_packets),
+                       kOpusSamplingKhz,
+                       kOpusSamplingKhz,
+                       SdpAudioFormat("opus", 48000, 2)),
+      opus_encoder_(NULL),
+      repacketizer_(NULL),
+      sub_block_size_samples_(
+          static_cast<size_t>(kOpusBlockDurationMs * kOpusSamplingKhz)),
+      bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)),
+      fec_(absl::GetFlag(FLAGS_fec)),
+      dtx_(absl::GetFlag(FLAGS_dtx)),
+      complexity_(absl::GetFlag(FLAGS_complexity)),
+      maxplaybackrate_(absl::GetFlag(FLAGS_maxplaybackrate)),
+      target_loss_rate_(absl::GetFlag(FLAGS_reported_loss_rate)),
+      sub_packets_(absl::GetFlag(FLAGS_sub_packets)) {
+  // Flag validation
+  RTC_CHECK(absl::GetFlag(FLAGS_bit_rate_kbps) >= 6 &&
+            absl::GetFlag(FLAGS_bit_rate_kbps) <= 510)
+      << "Invalid bit rate, should be between 6 and 510 kbps.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_complexity) >= -1 &&
+            absl::GetFlag(FLAGS_complexity) <= 10)
+      << "Invalid complexity setting, should be between 0 and 10.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_application) == 0 ||
+            absl::GetFlag(FLAGS_application) == 1)
+      << "Invalid application mode, should be 0 or 1.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_reported_loss_rate) >= 0 &&
+            absl::GetFlag(FLAGS_reported_loss_rate) <= 100)
+      << "Invalid packet loss percentile, should be between 0 and 100.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_sub_packets) >= 1 &&
+            absl::GetFlag(FLAGS_sub_packets) <= 3)
+      << "Invalid number of sub packets, should be between 1 and 3.";
+
+  // Redefine decoder type if input is stereo.
+  if (channels_ > 1) {
+    audio_format_ = SdpAudioFormat("opus", 48000, 2,
+                                   SdpAudioFormat::Parameters{{"stereo", "1"}});
+  }
+  application_ = absl::GetFlag(FLAGS_application);
+}
+
+void NetEqOpusQualityTest::SetUp() {
+  // Create encoder memory.
+  WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_, 48000);
+  ASSERT_TRUE(opus_encoder_);
+
+  // Create repacketizer.
+  repacketizer_ = opus_repacketizer_create();
+  ASSERT_TRUE(repacketizer_);
+
+  // Set bitrate.
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_kbps_ * 1000));
+  if (fec_) {
+    EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
+  }
+  if (dtx_) {
+    EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
+  }
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity_));
+  EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, maxplaybackrate_));
+  EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, target_loss_rate_));
+  NetEqQualityTest::SetUp();
+}
+
+void NetEqOpusQualityTest::TearDown() {
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  opus_repacketizer_destroy(repacketizer_);
+  NetEqQualityTest::TearDown();
+}
+
+int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
+                                      size_t block_size_samples,
+                                      rtc::Buffer* payload,
+                                      size_t max_bytes) {
+  EXPECT_EQ(block_size_samples, sub_block_size_samples_ * sub_packets_);
+  int16_t* pointer = in_data;
+  int value;
+  opus_repacketizer_init(repacketizer_);
+  for (int idx = 0; idx < sub_packets_; idx++) {
+    payload->AppendData(max_bytes, [&](rtc::ArrayView<uint8_t> payload) {
+      value = WebRtcOpus_Encode(opus_encoder_, pointer, sub_block_size_samples_,
+                                max_bytes, payload.data());
+
+      Log() << "Encoded a frame with Opus mode "
+            << (value == 0 ? 0 : payload[0] >> 3) << std::endl;
+
+      return (value >= 0) ? static_cast<size_t>(value) : 0;
+    });
+
+    if (OPUS_OK !=
+        opus_repacketizer_cat(repacketizer_, payload->data(), value)) {
+      opus_repacketizer_init(repacketizer_);
+      // If the repacketization fails, we discard this frame.
+      return 0;
+    }
+    pointer += sub_block_size_samples_ * channels_;
+  }
+  value = opus_repacketizer_out(repacketizer_, payload->data(),
+                                static_cast<opus_int32>(max_bytes));
+  EXPECT_GE(value, 0);
+  return value;
+}
+
+TEST_F(NetEqOpusQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
new file mode 100644
index 0000000000..c3e160cb66
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
@@ -0,0 +1,81 @@
+/*
+ *  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 <memory>
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 48;
+static const int kOutputSampleRateKhz = 48;
+}  // namespace
+
+class NetEqPcm16bQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqPcm16bQualityTest()
+      : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         SdpAudioFormat("l16", 48000, 1)) {
+    // Flag validation
+    RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) >= 10 &&
+              absl::GetFlag(FLAGS_frame_size_ms) <= 60 &&
+              (absl::GetFlag(FLAGS_frame_size_ms) % 10) == 0)
+        << "Invalid frame size, should be 10, 20, ..., 60 ms.";
+  }
+
+  void SetUp() override {
+    AudioEncoderPcm16B::Config config;
+    config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+    config.sample_rate_hz = 48000;
+    config.num_channels = channels_;
+    encoder_.reset(new AudioEncoderPcm16B(config));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 480;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  std::unique_ptr<AudioEncoderPcm16B> encoder_;
+};
+
+TEST_F(NetEqPcm16bQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
new file mode 100644
index 0000000000..d22170c623
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -0,0 +1,80 @@
+/*
+ *  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 <memory>
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 8;
+static const int kOutputSampleRateKhz = 8;
+}  // namespace
+
+class NetEqPcmuQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqPcmuQualityTest()
+      : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         SdpAudioFormat("pcmu", 8000, 1)) {
+    // Flag validation
+    RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) >= 10 &&
+              absl::GetFlag(FLAGS_frame_size_ms) <= 60 &&
+              (absl::GetFlag(FLAGS_frame_size_ms) % 10) == 0)
+        << "Invalid frame size, should be 10, 20, ..., 60 ms.";
+  }
+
+  void SetUp() override {
+    ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
+    AudioEncoderPcmU::Config config;
+    config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+    encoder_.reset(new AudioEncoderPcmU(config));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  std::unique_ptr<AudioEncoderPcmU> encoder_;
+};
+
+TEST_F(NetEqPcmuQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
new file mode 100644
index 0000000000..961f74ab66
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metric.h"
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::test::GetGlobalMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+// Runs a test with 10% packet losses and 10% clock drift, to exercise
+// both loss concealment and time-stretching code.
+TEST(NetEqPerformanceTest, 10_Pl_10_Drift) {
+  const int kSimulationTimeMs = 10000000;
+  const int kQuickSimulationTimeMs = 100000;
+  const int kLossPeriod = 10;  // Drop every 10th packet.
+  const double kDriftFactor = 0.1;
+  int64_t runtime = test::NetEqPerformanceTest::Run(
+      field_trial::IsEnabled("WebRTC-QuickPerfTest") ? kQuickSimulationTimeMs
+                                                     : kSimulationTimeMs,
+      kLossPeriod, kDriftFactor);
+  ASSERT_GT(runtime, 0);
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "neteq_performance", "10_pl_10_drift", runtime, Unit::kMilliseconds,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+// Runs a test with neither packet losses nor clock drift, to put
+// emphasis on the "good-weather" code path, which is presumably much
+// more lightweight.
+TEST(NetEqPerformanceTest, 0_Pl_0_Drift) {
+  const int kSimulationTimeMs = 10000000;
+  const int kQuickSimulationTimeMs = 100000;
+  const int kLossPeriod = 0;        // No losses.
+  const double kDriftFactor = 0.0;  // No clock drift.
+  int64_t runtime = test::NetEqPerformanceTest::Run(
+      field_trial::IsEnabled("WebRTC-QuickPerfTest") ? kQuickSimulationTimeMs
+                                                     : kSimulationTimeMs,
+      kLossPeriod, kDriftFactor);
+  ASSERT_GT(runtime, 0);
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "neteq_performance", "0_pl_0_drift", runtime, Unit::kMilliseconds,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
new file mode 100644
index 0000000000..a72b2009eb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
@@ -0,0 +1,58 @@
+/*
+ *  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 <stdio.h>
+
+#include <iostream>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
+#include "rtc_base/checks.h"
+
+// Define command line flags.
+ABSL_FLAG(int, runtime_ms, 10000, "Simulated runtime in ms.");
+ABSL_FLAG(int, lossrate, 10, "Packet lossrate; drop every N packets.");
+ABSL_FLAG(float, drift, 0.1f, "Clockdrift factor.");
+
+int main(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  std::string program_name = args[0];
+  std::string usage =
+      "Tool for measuring the speed of NetEq.\n"
+      "Usage: " +
+      program_name +
+      " [options]\n\n"
+      "  --runtime_ms=N         runtime in ms; default is 10000 ms\n"
+      "  --lossrate=N           drop every N packets; default is 10\n"
+      "  --drift=F              clockdrift factor between 0.0 and 1.0; "
+      "default is 0.1\n";
+  if (args.size() != 1) {
+    printf("%s", usage.c_str());
+    return 1;
+  }
+  RTC_CHECK_GT(absl::GetFlag(FLAGS_runtime_ms), 0);
+  RTC_CHECK_GE(absl::GetFlag(FLAGS_lossrate), 0);
+  RTC_CHECK(absl::GetFlag(FLAGS_drift) >= 0.0 &&
+            absl::GetFlag(FLAGS_drift) < 1.0);
+
+  int64_t result = webrtc::test::NetEqPerformanceTest::Run(
+      absl::GetFlag(FLAGS_runtime_ms), absl::GetFlag(FLAGS_lossrate),
+      absl::GetFlag(FLAGS_drift));
+  if (result <= 0) {
+    std::cout << "There was an error" << std::endl;
+    return -1;
+  }
+
+  std::cout << "Simulation done" << std::endl;
+  std::cout << "Runtime = " << result << " ms" << std::endl;
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc
new file mode 100644
index 0000000000..f5d50dc859
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc
@@ -0,0 +1,109 @@
+/*
+ *  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/neteq/test/result_sink.h"
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/message_digest.h"
+#include "rtc_base/string_encode.h"
+#include "test/gtest.h"
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#else
+#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
+             webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
+  stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
+  stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
+  stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
+  stats->set_expand_rate(stats_raw.expand_rate);
+  stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
+  stats->set_preemptive_rate(stats_raw.preemptive_rate);
+  stats->set_accelerate_rate(stats_raw.accelerate_rate);
+  stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
+  stats->set_secondary_discarded_rate(stats_raw.secondary_discarded_rate);
+  stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
+  stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
+  stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
+  stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
+}
+
+void AddMessage(FILE* file,
+                rtc::MessageDigest* digest,
+                absl::string_view message) {
+  int32_t size = message.length();
+  if (file)
+    ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
+  digest->Update(&size, sizeof(size));
+
+  if (file)
+    ASSERT_EQ(static_cast<size_t>(size),
+              fwrite(message.data(), sizeof(char), size, file));
+  digest->Update(message.data(), sizeof(char) * size);
+}
+
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
+
+ResultSink::ResultSink(absl::string_view output_file)
+    : output_fp_(nullptr),
+      digest_(rtc::MessageDigestFactory::Create(rtc::DIGEST_SHA_1)) {
+  if (!output_file.empty()) {
+    output_fp_ = fopen(std::string(output_file).c_str(), "wb");
+    EXPECT_TRUE(output_fp_ != NULL);
+  }
+}
+
+ResultSink::~ResultSink() {
+  if (output_fp_)
+    fclose(output_fp_);
+}
+
+void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  neteq_unittest::NetEqNetworkStatistics stats;
+  Convert(stats_raw, &stats);
+
+  std::string stats_string;
+  ASSERT_TRUE(stats.SerializeToString(&stats_string));
+  AddMessage(output_fp_, digest_.get(), stats_string);
+#else
+  FAIL() << "Writing to reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
+}
+
+void ResultSink::VerifyChecksum(absl::string_view checksum) {
+  std::string buffer;
+  buffer.resize(digest_->Size());
+  digest_->Finish(buffer.data(), buffer.size());
+  const std::string result = rtc::hex_encode(buffer);
+  if (checksum.size() == result.size()) {
+    EXPECT_EQ(checksum, result);
+  } else {
+    // Check result is one the '|'-separated checksums.
+    EXPECT_NE(checksum.find(result), absl::string_view::npos)
+        << result << " should be one of these:\n"
+        << checksum;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.h b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.h
new file mode 100644
index 0000000000..c6923d7a7f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.h
@@ -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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_
+#define MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_
+
+#include <cstdio>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/neteq/neteq.h"
+#include "rtc_base/message_digest.h"
+
+namespace webrtc {
+
+class ResultSink {
+ public:
+  explicit ResultSink(absl::string_view output_file);
+  ~ResultSink();
+
+  template <typename T>
+  void AddResult(const T* test_results, size_t length);
+
+  void AddResult(const NetEqNetworkStatistics& stats);
+
+  void VerifyChecksum(absl::string_view ref_check_sum);
+
+ private:
+  FILE* output_fp_;
+  std::unique_ptr<rtc::MessageDigest> digest_;
+};
+
+template <typename T>
+void ResultSink::AddResult(const T* test_results, size_t length) {
+  if (output_fp_) {
+    ASSERT_EQ(length, fwrite(test_results, sizeof(T), length, output_fp_));
+  }
+  digest_->Update(test_results, sizeof(T) * length);
+}
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.cc b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.cc
new file mode 100644
index 0000000000..b89be0608d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.cc
@@ -0,0 +1,216 @@
+/*
+ *  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/neteq/time_stretch.h"
+
+#include <algorithm>  // min, max
+#include <memory>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
+                                              size_t input_len,
+                                              bool fast_mode,
+                                              AudioMultiVector* output,
+                                              size_t* length_change_samples) {
+  // Pre-calculate common multiplication with `fs_mult_`.
+  size_t fs_mult_120 =
+      static_cast<size_t>(fs_mult_ * 120);  // Corresponds to 15 ms.
+
+  const int16_t* signal;
+  std::unique_ptr<int16_t[]> signal_array;
+  size_t signal_len;
+  if (num_channels_ == 1) {
+    signal = input;
+    signal_len = input_len;
+  } else {
+    // We want `signal` to be only the first channel of `input`, which is
+    // interleaved. Thus, we take the first sample, skip forward `num_channels`
+    // samples, and continue like that.
+    signal_len = input_len / num_channels_;
+    signal_array.reset(new int16_t[signal_len]);
+    signal = signal_array.get();
+    size_t j = kRefChannel;
+    for (size_t i = 0; i < signal_len; ++i) {
+      signal_array[i] = input[j];
+      j += num_channels_;
+    }
+  }
+
+  // Find maximum absolute value of input signal.
+  max_input_value_ = WebRtcSpl_MaxAbsValueW16(signal, signal_len);
+
+  // Downsample to 4 kHz sample rate and calculate auto-correlation.
+  DspHelper::DownsampleTo4kHz(signal, signal_len, kDownsampledLen,
+                              sample_rate_hz_, true /* compensate delay*/,
+                              downsampled_input_);
+  AutoCorrelation();
+
+  // Find the strongest correlation peak.
+  static const size_t kNumPeaks = 1;
+  size_t peak_index;
+  int16_t peak_value;
+  DspHelper::PeakDetection(auto_correlation_, kCorrelationLen, kNumPeaks,
+                           fs_mult_, &peak_index, &peak_value);
+  // Assert that `peak_index` stays within boundaries.
+  RTC_DCHECK_LE(peak_index, (2 * kCorrelationLen - 1) * fs_mult_);
+
+  // Compensate peak_index for displaced starting position. The displacement
+  // happens in AutoCorrelation(). Here, `kMinLag` is in the down-sampled 4 kHz
+  // domain, while the `peak_index` is in the original sample rate; hence, the
+  // multiplication by fs_mult_ * 2.
+  peak_index += kMinLag * fs_mult_ * 2;
+  // Assert that `peak_index` stays within boundaries.
+  RTC_DCHECK_GE(peak_index, static_cast<size_t>(20 * fs_mult_));
+  RTC_DCHECK_LE(peak_index,
+                20 * fs_mult_ + (2 * kCorrelationLen - 1) * fs_mult_);
+
+  // Calculate scaling to ensure that `peak_index` samples can be square-summed
+  // without overflowing.
+  int scaling = 31 - WebRtcSpl_NormW32(max_input_value_ * max_input_value_) -
+                WebRtcSpl_NormW32(static_cast<int32_t>(peak_index));
+  scaling = std::max(0, scaling);
+
+  // `vec1` starts at 15 ms minus one pitch period.
+  const int16_t* vec1 = &signal[fs_mult_120 - peak_index];
+  // `vec2` start at 15 ms.
+  const int16_t* vec2 = &signal[fs_mult_120];
+  // Calculate energies for `vec1` and `vec2`, assuming they both contain
+  // `peak_index` samples.
+  int32_t vec1_energy =
+      WebRtcSpl_DotProductWithScale(vec1, vec1, peak_index, scaling);
+  int32_t vec2_energy =
+      WebRtcSpl_DotProductWithScale(vec2, vec2, peak_index, scaling);
+
+  // Calculate cross-correlation between `vec1` and `vec2`.
+  int32_t cross_corr =
+      WebRtcSpl_DotProductWithScale(vec1, vec2, peak_index, scaling);
+
+  // Check if the signal seems to be active speech or not (simple VAD).
+  bool active_speech =
+      SpeechDetection(vec1_energy, vec2_energy, peak_index, scaling);
+
+  int16_t best_correlation;
+  if (!active_speech) {
+    SetParametersForPassiveSpeech(signal_len, &best_correlation, &peak_index);
+  } else {
+    // Calculate correlation:
+    // cross_corr / sqrt(vec1_energy * vec2_energy).
+
+    // Start with calculating scale values.
+    int energy1_scale = std::max(0, 16 - WebRtcSpl_NormW32(vec1_energy));
+    int energy2_scale = std::max(0, 16 - WebRtcSpl_NormW32(vec2_energy));
+
+    // Make sure total scaling is even (to simplify scale factor after sqrt).
+    if ((energy1_scale + energy2_scale) & 1) {
+      // The sum is odd.
+      energy1_scale += 1;
+    }
+
+    // Scale energies to int16_t.
+    int16_t vec1_energy_int16 =
+        static_cast<int16_t>(vec1_energy >> energy1_scale);
+    int16_t vec2_energy_int16 =
+        static_cast<int16_t>(vec2_energy >> energy2_scale);
+
+    // Calculate square-root of energy product.
+    int16_t sqrt_energy_prod =
+        WebRtcSpl_SqrtFloor(vec1_energy_int16 * vec2_energy_int16);
+
+    // Calculate cross_corr / sqrt(en1*en2) in Q14.
+    int temp_scale = 14 - (energy1_scale + energy2_scale) / 2;
+    cross_corr = WEBRTC_SPL_SHIFT_W32(cross_corr, temp_scale);
+    cross_corr = std::max(0, cross_corr);  // Don't use if negative.
+    best_correlation = WebRtcSpl_DivW32W16(cross_corr, sqrt_energy_prod);
+    // Make sure `best_correlation` is no larger than 1 in Q14.
+    best_correlation = std::min(static_cast<int16_t>(16384), best_correlation);
+  }
+
+  // Check accelerate criteria and stretch the signal.
+  ReturnCodes return_value =
+      CheckCriteriaAndStretch(input, input_len, peak_index, best_correlation,
+                              active_speech, fast_mode, output);
+  switch (return_value) {
+    case kSuccess:
+      *length_change_samples = peak_index;
+      break;
+    case kSuccessLowEnergy:
+      *length_change_samples = peak_index;
+      break;
+    case kNoStretch:
+    case kError:
+      *length_change_samples = 0;
+      break;
+  }
+  return return_value;
+}
+
+void TimeStretch::AutoCorrelation() {
+  // Calculate correlation from lag kMinLag to lag kMaxLag in 4 kHz domain.
+  int32_t auto_corr[kCorrelationLen];
+  CrossCorrelationWithAutoShift(
+      &downsampled_input_[kMaxLag], &downsampled_input_[kMaxLag - kMinLag],
+      kCorrelationLen, kMaxLag - kMinLag, -1, auto_corr);
+
+  // Normalize correlation to 14 bits and write to `auto_correlation_`.
+  int32_t max_corr = WebRtcSpl_MaxAbsValueW32(auto_corr, kCorrelationLen);
+  int scaling = std::max(0, 17 - WebRtcSpl_NormW32(max_corr));
+  WebRtcSpl_VectorBitShiftW32ToW16(auto_correlation_, kCorrelationLen,
+                                   auto_corr, scaling);
+}
+
+bool TimeStretch::SpeechDetection(int32_t vec1_energy,
+                                  int32_t vec2_energy,
+                                  size_t peak_index,
+                                  int scaling) const {
+  // Check if the signal seems to be active speech or not (simple VAD).
+  // If (vec1_energy + vec2_energy) / (2 * peak_index) <=
+  // 8 * background_noise_energy, then we say that the signal contains no
+  // active speech.
+  // Rewrite the inequality as:
+  // (vec1_energy + vec2_energy) / 16 <= peak_index * background_noise_energy.
+  // The two sides of the inequality will be denoted `left_side` and
+  // `right_side`.
+  int32_t left_side = rtc::saturated_cast<int32_t>(
+      (static_cast<int64_t>(vec1_energy) + vec2_energy) / 16);
+  int32_t right_side;
+  if (background_noise_.initialized()) {
+    right_side = background_noise_.Energy(kRefChannel);
+  } else {
+    // If noise parameters have not been estimated, use a fixed threshold.
+    right_side = 75000;
+  }
+  int right_scale = 16 - WebRtcSpl_NormW32(right_side);
+  right_scale = std::max(0, right_scale);
+  left_side = left_side >> right_scale;
+  right_side =
+      rtc::dchecked_cast<int32_t>(peak_index) * (right_side >> right_scale);
+
+  // Scale `left_side` properly before comparing with `right_side`.
+  // (`scaling` is the scale factor before energy calculation, thus the scale
+  // factor for the energy is 2 * scaling.)
+  if (WebRtcSpl_NormW32(left_side) < 2 * scaling) {
+    // Cannot scale only `left_side`, must scale `right_side` too.
+    int temp_scale = WebRtcSpl_NormW32(left_side);
+    left_side = left_side << temp_scale;
+    right_side = right_side >> (2 * scaling - temp_scale);
+  } else {
+    left_side = left_side << 2 * scaling;
+  }
+  return left_side > right_side;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.h b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.h
new file mode 100644
index 0000000000..f0ddaebeca
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch.h
@@ -0,0 +1,113 @@
+/*
+ *  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_NETEQ_TIME_STRETCH_H_
+#define MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
+
+#include <string.h>  // memset, size_t
+
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class BackgroundNoise;
+
+// This is the base class for Accelerate and PreemptiveExpand. This class
+// cannot be instantiated, but must be used through either of the derived
+// classes.
+class TimeStretch {
+ public:
+  enum ReturnCodes {
+    kSuccess = 0,
+    kSuccessLowEnergy = 1,
+    kNoStretch = 2,
+    kError = -1
+  };
+
+  TimeStretch(int sample_rate_hz,
+              size_t num_channels,
+              const BackgroundNoise& background_noise)
+      : sample_rate_hz_(sample_rate_hz),
+        fs_mult_(sample_rate_hz / 8000),
+        num_channels_(num_channels),
+        background_noise_(background_noise),
+        max_input_value_(0) {
+    RTC_DCHECK(sample_rate_hz_ == 8000 || sample_rate_hz_ == 16000 ||
+               sample_rate_hz_ == 32000 || sample_rate_hz_ == 48000);
+    RTC_DCHECK_GT(num_channels_, 0);
+    memset(auto_correlation_, 0, sizeof(auto_correlation_));
+  }
+
+  virtual ~TimeStretch() {}
+
+  TimeStretch(const TimeStretch&) = delete;
+  TimeStretch& operator=(const TimeStretch&) = delete;
+
+  // This method performs the processing common to both Accelerate and
+  // PreemptiveExpand.
+  ReturnCodes Process(const int16_t* input,
+                      size_t input_len,
+                      bool fast_mode,
+                      AudioMultiVector* output,
+                      size_t* length_change_samples);
+
+ protected:
+  // Sets the parameters `best_correlation` and `peak_index` to suitable
+  // values when the signal contains no active speech. This method must be
+  // implemented by the sub-classes.
+  virtual void SetParametersForPassiveSpeech(size_t input_length,
+                                             int16_t* best_correlation,
+                                             size_t* peak_index) const = 0;
+
+  // Checks the criteria for performing the time-stretching operation and,
+  // if possible, performs the time-stretching. This method must be implemented
+  // by the sub-classes.
+  virtual ReturnCodes CheckCriteriaAndStretch(
+      const int16_t* input,
+      size_t input_length,
+      size_t peak_index,
+      int16_t best_correlation,
+      bool active_speech,
+      bool fast_mode,
+      AudioMultiVector* output) const = 0;
+
+  static const size_t kCorrelationLen = 50;
+  static const size_t kLogCorrelationLen = 6;  // >= log2(kCorrelationLen).
+  static const size_t kMinLag = 10;
+  static const size_t kMaxLag = 60;
+  static const size_t kDownsampledLen = kCorrelationLen + kMaxLag;
+  static const int kCorrelationThreshold = 14746;  // 0.9 in Q14.
+  static constexpr size_t kRefChannel = 0;  // First channel is reference.
+
+  const int sample_rate_hz_;
+  const int fs_mult_;  // Sample rate multiplier = sample_rate_hz_ / 8000.
+  const size_t num_channels_;
+  const BackgroundNoise& background_noise_;
+  int16_t max_input_value_;
+  int16_t downsampled_input_[kDownsampledLen];
+  // Adding 1 to the size of `auto_correlation_` because of how it is used
+  // by the peak-detection algorithm.
+  int16_t auto_correlation_[kCorrelationLen + 1];
+
+ private:
+  // Calculates the auto-correlation of `downsampled_input_` and writes the
+  // result to `auto_correlation_`.
+  void AutoCorrelation();
+
+  // Performs a simple voice-activity detection based on the input parameters.
+  bool SpeechDetection(int32_t vec1_energy,
+                       int32_t vec2_energy,
+                       size_t peak_index,
+                       int scaling) const;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
new file mode 100644
index 0000000000..da3a98229a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
@@ -0,0 +1,124 @@
+/*
+ *  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.
+ */
+
+// Unit tests for Accelerate and PreemptiveExpand classes.
+
+#include <map>
+#include <memory>
+
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/accelerate.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace {
+const size_t kNumChannels = 1;
+}
+
+TEST(TimeStretch, CreateAndDestroy) {
+  const int kSampleRate = 8000;
+  const int kOverlapSamples = 5 * kSampleRate / 8000;
+  BackgroundNoise bgn(kNumChannels);
+  Accelerate accelerate(kSampleRate, kNumChannels, bgn);
+  PreemptiveExpand preemptive_expand(kSampleRate, kNumChannels, bgn,
+                                     kOverlapSamples);
+}
+
+TEST(TimeStretch, CreateUsingFactory) {
+  const int kSampleRate = 8000;
+  const int kOverlapSamples = 5 * kSampleRate / 8000;
+  BackgroundNoise bgn(kNumChannels);
+
+  AccelerateFactory accelerate_factory;
+  Accelerate* accelerate =
+      accelerate_factory.Create(kSampleRate, kNumChannels, bgn);
+  EXPECT_TRUE(accelerate != NULL);
+  delete accelerate;
+
+  PreemptiveExpandFactory preemptive_expand_factory;
+  PreemptiveExpand* preemptive_expand = preemptive_expand_factory.Create(
+      kSampleRate, kNumChannels, bgn, kOverlapSamples);
+  EXPECT_TRUE(preemptive_expand != NULL);
+  delete preemptive_expand;
+}
+
+class TimeStretchTest : public ::testing::Test {
+ protected:
+  TimeStretchTest()
+      : input_file_(new test::InputAudioFile(
+            test::ResourcePath("audio_coding/testfile32kHz", "pcm"))),
+        sample_rate_hz_(32000),
+        block_size_(30 * sample_rate_hz_ / 1000),  // 30 ms
+        audio_(new int16_t[block_size_]),
+        background_noise_(kNumChannels) {}
+
+  const int16_t* Next30Ms() {
+    RTC_CHECK(input_file_->Read(block_size_, audio_.get()));
+    return audio_.get();
+  }
+
+  // Returns the total length change (in samples) that the accelerate operation
+  // resulted in during the run.
+  size_t TestAccelerate(size_t loops, bool fast_mode) {
+    Accelerate accelerate(sample_rate_hz_, kNumChannels, background_noise_);
+    size_t total_length_change = 0;
+    for (size_t i = 0; i < loops; ++i) {
+      AudioMultiVector output(kNumChannels);
+      size_t length_change;
+      UpdateReturnStats(accelerate.Process(Next30Ms(), block_size_, fast_mode,
+                                           &output, &length_change));
+      total_length_change += length_change;
+    }
+    return total_length_change;
+  }
+
+  void UpdateReturnStats(TimeStretch::ReturnCodes ret) {
+    switch (ret) {
+      case TimeStretch::kSuccess:
+      case TimeStretch::kSuccessLowEnergy:
+      case TimeStretch::kNoStretch:
+        ++return_stats_[ret];
+        break;
+      case TimeStretch::kError:
+        FAIL() << "Process returned an error";
+    }
+  }
+
+  std::unique_ptr<test::InputAudioFile> input_file_;
+  const int sample_rate_hz_;
+  const size_t block_size_;
+  std::unique_ptr<int16_t[]> audio_;
+  std::map<TimeStretch::ReturnCodes, int> return_stats_;
+  BackgroundNoise background_noise_;
+};
+
+TEST_F(TimeStretchTest, Accelerate) {
+  // TestAccelerate returns the total length change in samples.
+  EXPECT_EQ(15268U, TestAccelerate(100, false));
+  EXPECT_EQ(9, return_stats_[TimeStretch::kSuccess]);
+  EXPECT_EQ(58, return_stats_[TimeStretch::kSuccessLowEnergy]);
+  EXPECT_EQ(33, return_stats_[TimeStretch::kNoStretch]);
+}
+
+TEST_F(TimeStretchTest, AccelerateFastMode) {
+  // TestAccelerate returns the total length change in samples.
+  EXPECT_EQ(21400U, TestAccelerate(100, true));
+  EXPECT_EQ(31, return_stats_[TimeStretch::kSuccess]);
+  EXPECT_EQ(58, return_stats_[TimeStretch::kSuccessLowEnergy]);
+  EXPECT_EQ(11, return_stats_[TimeStretch::kNoStretch]);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.cc b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.cc
new file mode 100644
index 0000000000..59177d027f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.cc
@@ -0,0 +1,87 @@
+/*
+ *  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/neteq/timestamp_scaler.h"
+
+#include "api/audio_codecs/audio_format.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void TimestampScaler::Reset() {
+  first_packet_received_ = false;
+}
+
+void TimestampScaler::ToInternal(Packet* packet) {
+  if (!packet) {
+    return;
+  }
+  packet->timestamp = ToInternal(packet->timestamp, packet->payload_type);
+}
+
+void TimestampScaler::ToInternal(PacketList* packet_list) {
+  PacketList::iterator it;
+  for (it = packet_list->begin(); it != packet_list->end(); ++it) {
+    ToInternal(&(*it));
+  }
+}
+
+uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
+                                     uint8_t rtp_payload_type) {
+  const DecoderDatabase::DecoderInfo* info =
+      decoder_database_.GetDecoderInfo(rtp_payload_type);
+  if (!info) {
+    // Payload type is unknown. Do not scale.
+    return external_timestamp;
+  }
+  if (!(info->IsComfortNoise() || info->IsDtmf())) {
+    // Do not change the timestamp scaling settings for DTMF or CNG.
+    numerator_ = info->SampleRateHz();
+    if (info->GetFormat().clockrate_hz == 0) {
+      // If the clockrate is invalid (i.e. with an old-style external codec)
+      // we cannot do any timestamp scaling.
+      denominator_ = numerator_;
+    } else {
+      denominator_ = info->GetFormat().clockrate_hz;
+    }
+  }
+  if (numerator_ != denominator_) {
+    // We have a scale factor != 1.
+    if (!first_packet_received_) {
+      external_ref_ = external_timestamp;
+      internal_ref_ = external_timestamp;
+      first_packet_received_ = true;
+    }
+    const int64_t external_diff = int64_t{external_timestamp} - external_ref_;
+    RTC_DCHECK_GT(denominator_, 0);
+    external_ref_ = external_timestamp;
+    internal_ref_ += (external_diff * numerator_) / denominator_;
+    return internal_ref_;
+  } else {
+    // No scaling.
+    return external_timestamp;
+  }
+}
+
+uint32_t TimestampScaler::ToExternal(uint32_t internal_timestamp) const {
+  if (!first_packet_received_ || (numerator_ == denominator_)) {
+    // Not initialized, or scale factor is 1.
+    return internal_timestamp;
+  } else {
+    const int64_t internal_diff = int64_t{internal_timestamp} - internal_ref_;
+    RTC_DCHECK_GT(numerator_, 0);
+    // Do not update references in this method.
+    // Switch `denominator_` and `numerator_` to convert the other way.
+    return external_ref_ + (internal_diff * denominator_) / numerator_;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.h b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.h
new file mode 100644
index 0000000000..f42ce7207a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler.h
@@ -0,0 +1,67 @@
+/*
+ *  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_NETEQ_TIMESTAMP_SCALER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
+
+#include "modules/audio_coding/neteq/packet.h"
+
+namespace webrtc {
+
+// Forward declaration.
+class DecoderDatabase;
+
+// This class scales timestamps for codecs that need timestamp scaling.
+// This is done for codecs where one RTP timestamp does not correspond to
+// one sample.
+class TimestampScaler {
+ public:
+  explicit TimestampScaler(const DecoderDatabase& decoder_database)
+      : first_packet_received_(false),
+        numerator_(1),
+        denominator_(1),
+        external_ref_(0),
+        internal_ref_(0),
+        decoder_database_(decoder_database) {}
+
+  virtual ~TimestampScaler() {}
+
+  TimestampScaler(const TimestampScaler&) = delete;
+  TimestampScaler& operator=(const TimestampScaler&) = delete;
+
+  // Start over.
+  virtual void Reset();
+
+  // Scale the timestamp in `packet` from external to internal.
+  virtual void ToInternal(Packet* packet);
+
+  // Scale the timestamp for all packets in `packet_list` from external to
+  // internal.
+  virtual void ToInternal(PacketList* packet_list);
+
+  // Returns the internal equivalent of `external_timestamp`, given the
+  // RTP payload type `rtp_payload_type`.
+  virtual uint32_t ToInternal(uint32_t external_timestamp,
+                              uint8_t rtp_payload_type);
+
+  // Scales back to external timestamp. This is the inverse of ToInternal().
+  virtual uint32_t ToExternal(uint32_t internal_timestamp) const;
+
+ private:
+  bool first_packet_received_;
+  int numerator_;
+  int denominator_;
+  uint32_t external_ref_;
+  uint32_t internal_ref_;
+  const DecoderDatabase& decoder_database_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
new file mode 100644
index 0000000000..c2bb4dd95f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
@@ -0,0 +1,324 @@
+/*
+ *  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/neteq/timestamp_scaler.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::Return;
+using ::testing::ReturnNull;
+
+namespace webrtc {
+
+TEST(TimestampScaler, TestNoScaling) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use PCMu, because it doesn't use scaled timestamps.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 0;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  for (uint32_t timestamp = 0xFFFFFFFF - 5; timestamp != 5; ++timestamp) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(timestamp, scaler.ToInternal(timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(timestamp, scaler.ToExternal(timestamp));
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, TestNoScalingLargeStep) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use PCMu, because it doesn't use scaled timestamps.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 0;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  static const uint32_t kStep = 160;
+  uint32_t start_timestamp = 0;
+  // `external_timestamp` will be a large positive value.
+  start_timestamp = start_timestamp - 5 * kStep;
+  for (uint32_t timestamp = start_timestamp; timestamp != 5 * kStep;
+       timestamp += kStep) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(timestamp, scaler.ToInternal(timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(timestamp, scaler.ToExternal(timestamp));
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, TestG722) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  uint32_t external_timestamp = 0xFFFFFFFF - 5;
+  uint32_t internal_timestamp = external_timestamp;
+  for (; external_timestamp != 5; ++external_timestamp) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(internal_timestamp,
+              scaler.ToInternal(external_timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    internal_timestamp += 2;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, TestG722LargeStep) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  static const uint32_t kStep = 320;
+  uint32_t external_timestamp = 0;
+  // `external_timestamp` will be a large positive value.
+  external_timestamp = external_timestamp - 5 * kStep;
+  uint32_t internal_timestamp = external_timestamp;
+  for (; external_timestamp != 5 * kStep; external_timestamp += kStep) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(internal_timestamp,
+              scaler.ToInternal(external_timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    // Internal timestamp should be incremented with twice the step.
+    internal_timestamp += 2 * kStep;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, TestG722WithCng) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info_g722(SdpAudioFormat("g722", 8000, 1),
+                                               absl::nullopt, factory.get());
+  const DecoderDatabase::DecoderInfo info_cng(SdpAudioFormat("cn", 16000, 1),
+                                              absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadTypeG722 = 17;
+  static const uint8_t kRtpPayloadTypeCng = 13;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadTypeG722))
+      .WillRepeatedly(Return(&info_g722));
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadTypeCng))
+      .WillRepeatedly(Return(&info_cng));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  uint32_t external_timestamp = 0xFFFFFFFF - 5;
+  uint32_t internal_timestamp = external_timestamp;
+  bool next_is_cng = false;
+  for (; external_timestamp != 5; ++external_timestamp) {
+    // Alternate between G.722 and CNG every other packet.
+    if (next_is_cng) {
+      // Scale to internal timestamp.
+      EXPECT_EQ(internal_timestamp,
+                scaler.ToInternal(external_timestamp, kRtpPayloadTypeCng));
+      next_is_cng = false;
+    } else {
+      // Scale to internal timestamp.
+      EXPECT_EQ(internal_timestamp,
+                scaler.ToInternal(external_timestamp, kRtpPayloadTypeG722));
+      next_is_cng = true;
+    }
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    internal_timestamp += 2;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+// Make sure that the method ToInternal(Packet* packet) is wired up correctly.
+// Since it is simply calling the other ToInternal method, we are not doing
+// as many tests here.
+TEST(TimestampScaler, TestG722Packet) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  uint32_t external_timestamp = 0xFFFFFFFF - 5;
+  uint32_t internal_timestamp = external_timestamp;
+  Packet packet;
+  packet.payload_type = kRtpPayloadType;
+  for (; external_timestamp != 5; ++external_timestamp) {
+    packet.timestamp = external_timestamp;
+    // Scale to internal timestamp.
+    scaler.ToInternal(&packet);
+    EXPECT_EQ(internal_timestamp, packet.timestamp);
+    internal_timestamp += 2;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+// Make sure that the method ToInternal(PacketList* packet_list) is wired up
+// correctly. Since it is simply calling the ToInternal(Packet* packet) method,
+// we are not doing as many tests here.
+TEST(TimestampScaler, TestG722PacketList) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  uint32_t external_timestamp = 0xFFFFFFFF - 5;
+  uint32_t internal_timestamp = external_timestamp;
+  PacketList packet_list;
+  {
+    Packet packet1;
+    packet1.payload_type = kRtpPayloadType;
+    packet1.timestamp = external_timestamp;
+    Packet packet2;
+    packet2.payload_type = kRtpPayloadType;
+    packet2.timestamp = external_timestamp + 10;
+    packet_list.push_back(std::move(packet1));
+    packet_list.push_back(std::move(packet2));
+  }
+
+  scaler.ToInternal(&packet_list);
+  EXPECT_EQ(internal_timestamp, packet_list.front().timestamp);
+  packet_list.pop_front();
+  EXPECT_EQ(internal_timestamp + 20, packet_list.front().timestamp);
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, TestG722Reset) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  // Use G722, which has a factor 2 scaling.
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("g722", 8000, 1),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  uint32_t external_timestamp = 0xFFFFFFFF - 5;
+  uint32_t internal_timestamp = external_timestamp;
+  for (; external_timestamp != 5; ++external_timestamp) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(internal_timestamp,
+              scaler.ToInternal(external_timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    internal_timestamp += 2;
+  }
+  // Reset the scaler. After this, we expect the internal and external to start
+  // over at the same value again.
+  scaler.Reset();
+  internal_timestamp = external_timestamp;
+  for (; external_timestamp != 15; ++external_timestamp) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(internal_timestamp,
+              scaler.ToInternal(external_timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    internal_timestamp += 2;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+// TODO(minyue): This test becomes trivial since Opus does not need a timestamp
+// scaler. Therefore, this test may be removed in future. There is no harm to
+// keep it, since it can be taken as a test case for the situation of a trivial
+// timestamp scaler.
+TEST(TimestampScaler, TestOpusLargeStep) {
+  MockDecoderDatabase db;
+  auto factory = CreateBuiltinAudioDecoderFactory();
+  const DecoderDatabase::DecoderInfo info(SdpAudioFormat("opus", 48000, 2),
+                                          absl::nullopt, factory.get());
+  static const uint8_t kRtpPayloadType = 17;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillRepeatedly(Return(&info));
+
+  TimestampScaler scaler(db);
+  // Test both sides of the timestamp wrap-around.
+  static const uint32_t kStep = 960;
+  uint32_t external_timestamp = 0;
+  // `external_timestamp` will be a large positive value.
+  external_timestamp = external_timestamp - 5 * kStep;
+  uint32_t internal_timestamp = external_timestamp;
+  for (; external_timestamp != 5 * kStep; external_timestamp += kStep) {
+    // Scale to internal timestamp.
+    EXPECT_EQ(internal_timestamp,
+              scaler.ToInternal(external_timestamp, kRtpPayloadType));
+    // Scale back.
+    EXPECT_EQ(external_timestamp, scaler.ToExternal(internal_timestamp));
+    internal_timestamp += kStep;
+  }
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+TEST(TimestampScaler, Failures) {
+  static const uint8_t kRtpPayloadType = 17;
+  MockDecoderDatabase db;
+  EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
+      .WillOnce(ReturnNull());  // Return NULL to indicate unknown payload type.
+
+  TimestampScaler scaler(db);
+  uint32_t timestamp = 4711;  // Some number.
+  EXPECT_EQ(timestamp, scaler.ToInternal(timestamp, kRtpPayloadType));
+
+  Packet* packet = NULL;
+  scaler.ToInternal(packet);  // Should not crash. That's all we can test.
+
+  EXPECT_CALL(db, Die());  // Called when database object is deleted.
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/DEPS b/third_party/libwebrtc/modules/audio_coding/neteq/tools/DEPS
new file mode 100644
index 0000000000..4db1e1d6e5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/DEPS
@@ -0,0 +1,3 @@
+include_rules = [
+  "+logging/rtc_event_log",
+]
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/README.md b/third_party/libwebrtc/modules/audio_coding/neteq/tools/README.md
new file mode 100644
index 0000000000..e7bd95c285
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/README.md
@@ -0,0 +1,17 @@
+# NetEQ RTP Play tool
+
+## Testing of the command line arguments
+The command line tool `neteq_rtpplay` can be tested by running `neteq_rtpplay_test.sh`, which is not use on try bots, but it can be used before submitting any CLs that may break the behavior of the command line arguments of `neteq_rtpplay`.
+
+Run `neteq_rtpplay_test.sh` as follows from the `src/` folder:
+```
+src$ ./modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh  \
+  out/Default/neteq_rtpplay  \
+  resources/audio_coding/neteq_opus.rtp  \
+  resources/short_mixed_mono_48.pcm
+```
+
+You can replace the RTP and PCM files with any other compatible files.
+If you get an error using the files indicated above, try running `gclient sync`.
+
+Requirements: `awk` and `md5sum`.
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_checksum.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_checksum.h
new file mode 100644
index 0000000000..42e3a3a3a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -0,0 +1,64 @@
+/*
+ *  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_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+
+#include <memory>
+#include <string>
+
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/message_digest.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace test {
+
+class AudioChecksum : public AudioSink {
+ public:
+  AudioChecksum()
+      : checksum_(rtc::MessageDigestFactory::Create(rtc::DIGEST_MD5)),
+        checksum_result_(checksum_->Size()),
+        finished_(false) {}
+
+  AudioChecksum(const AudioChecksum&) = delete;
+  AudioChecksum& operator=(const AudioChecksum&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
+    if (finished_)
+      return false;
+
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+#error "Big-endian gives a different checksum"
+#endif
+    checksum_->Update(audio, num_samples * sizeof(*audio));
+    return true;
+  }
+
+  // Finalizes the computations, and returns the checksum.
+  std::string Finish() {
+    if (!finished_) {
+      finished_ = true;
+      checksum_->Finish(checksum_result_.data(), checksum_result_.size());
+    }
+    return rtc::hex_encode(checksum_result_);
+  }
+
+ private:
+  std::unique_ptr<rtc::MessageDigest> checksum_;
+  rtc::Buffer checksum_result_;
+  bool finished_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.cc
new file mode 100644
index 0000000000..514e6eb2ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.cc
@@ -0,0 +1,61 @@
+/*
+ *  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 "modules/audio_coding/neteq/tools/audio_loop.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+bool AudioLoop::Init(absl::string_view file_name,
+                     size_t max_loop_length_samples,
+                     size_t block_length_samples) {
+  FILE* fp = fopen(std::string(file_name).c_str(), "rb");
+  if (!fp)
+    return false;
+
+  audio_array_.reset(
+      new int16_t[max_loop_length_samples + block_length_samples]);
+  size_t samples_read =
+      fread(audio_array_.get(), sizeof(int16_t), max_loop_length_samples, fp);
+  fclose(fp);
+
+  // Block length must be shorter than the loop length.
+  if (block_length_samples > samples_read)
+    return false;
+
+  // 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(&audio_array_[samples_read], audio_array_.get(),
+         block_length_samples * sizeof(int16_t));
+
+  loop_length_samples_ = samples_read;
+  block_length_samples_ = block_length_samples;
+  next_index_ = 0;
+  return true;
+}
+
+rtc::ArrayView<const int16_t> AudioLoop::GetNextBlock() {
+  // Check that the AudioLoop is initialized.
+  if (block_length_samples_ == 0)
+    return rtc::ArrayView<const int16_t>();
+
+  const int16_t* output_ptr = &audio_array_[next_index_];
+  next_index_ = (next_index_ + block_length_samples_) % loop_length_samples_;
+  return rtc::ArrayView<const int16_t>(output_ptr, block_length_samples_);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.h
new file mode 100644
index 0000000000..f5f0b59011
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_loop.h
@@ -0,0 +1,57 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+
+namespace webrtc {
+namespace test {
+
+// Class serving as an infinite source of audio, realized by looping an audio
+// clip.
+class AudioLoop {
+ public:
+  AudioLoop()
+      : next_index_(0), loop_length_samples_(0), block_length_samples_(0) {}
+
+  virtual ~AudioLoop() {}
+
+  AudioLoop(const AudioLoop&) = delete;
+  AudioLoop& operator=(const AudioLoop&) = delete;
+
+  // Initializes the AudioLoop by reading from `file_name`. The loop will be no
+  // longer than `max_loop_length_samples`, if the length of the file is
+  // greater. Otherwise, the loop length is the same as the file length.
+  // The audio will be delivered in blocks of `block_length_samples`.
+  // Returns false if the initialization failed, otherwise true.
+  bool Init(absl::string_view file_name,
+            size_t max_loop_length_samples,
+            size_t block_length_samples);
+
+  // Returns a (pointer,size) pair for the next block of audio. The size is
+  // equal to the `block_length_samples` Init() argument.
+  rtc::ArrayView<const int16_t> GetNextBlock();
+
+ private:
+  size_t next_index_;
+  size_t loop_length_samples_;
+  size_t block_length_samples_;
+  std::unique_ptr<int16_t[]> audio_array_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.cc
new file mode 100644
index 0000000000..7d7af7ef9f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.cc
@@ -0,0 +1,26 @@
+/*
+ *  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/neteq/tools/audio_sink.h"
+
+namespace webrtc {
+namespace test {
+
+bool AudioSinkFork::WriteArray(const int16_t* audio, size_t num_samples) {
+  return left_sink_->WriteArray(audio, num_samples) &&
+         right_sink_->WriteArray(audio, num_samples);
+}
+
+bool VoidAudioSink::WriteArray(const int16_t* audio, size_t num_samples) {
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.h
new file mode 100644
index 0000000000..53729fa920
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/audio_sink.h
@@ -0,0 +1,70 @@
+/*
+ *  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_NETEQ_TOOLS_AUDIO_SINK_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
+
+#include "api/audio/audio_frame.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for an object receiving raw output audio from test
+// applications.
+class AudioSink {
+ public:
+  AudioSink() {}
+  virtual ~AudioSink() {}
+
+  AudioSink(const AudioSink&) = delete;
+  AudioSink& operator=(const AudioSink&) = delete;
+
+  // Writes `num_samples` from `audio` to the AudioSink. Returns true if
+  // successful, otherwise false.
+  virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
+
+  // Writes `audio_frame` to the AudioSink. Returns true if successful,
+  // otherwise false.
+  bool WriteAudioFrame(const AudioFrame& audio_frame) {
+    return WriteArray(audio_frame.data(), audio_frame.samples_per_channel_ *
+                                              audio_frame.num_channels_);
+  }
+};
+
+// Forks the output audio to two AudioSink objects.
+class AudioSinkFork : public AudioSink {
+ public:
+  AudioSinkFork(AudioSink* left, AudioSink* right)
+      : left_sink_(left), right_sink_(right) {}
+
+  AudioSinkFork(const AudioSinkFork&) = delete;
+  AudioSinkFork& operator=(const AudioSinkFork&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override;
+
+ private:
+  AudioSink* left_sink_;
+  AudioSink* right_sink_;
+};
+
+// An AudioSink implementation that does nothing.
+class VoidAudioSink : public AudioSink {
+ public:
+  VoidAudioSink() = default;
+
+  VoidAudioSink(const VoidAudioSink&) = delete;
+  VoidAudioSink& operator=(const VoidAudioSink&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
new file mode 100644
index 0000000000..18a910365f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
@@ -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.
+ */
+
+#include "modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
+
+#include <algorithm>
+
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+ConstantPcmPacketSource::ConstantPcmPacketSource(size_t payload_len_samples,
+                                                 int16_t sample_value,
+                                                 int sample_rate_hz,
+                                                 int payload_type)
+    : payload_len_samples_(payload_len_samples),
+      packet_len_bytes_(2 * payload_len_samples_ + kHeaderLenBytes),
+      samples_per_ms_(sample_rate_hz / 1000),
+      next_arrival_time_ms_(0.0),
+      payload_type_(payload_type),
+      seq_number_(0),
+      timestamp_(0),
+      payload_ssrc_(0xABCD1234) {
+  size_t encoded_len = WebRtcPcm16b_Encode(&sample_value, 1, encoded_sample_);
+  RTC_CHECK_EQ(2U, encoded_len);
+}
+
+std::unique_ptr<Packet> ConstantPcmPacketSource::NextPacket() {
+  RTC_CHECK_GT(packet_len_bytes_, kHeaderLenBytes);
+  rtc::CopyOnWriteBuffer packet_buffer(packet_len_bytes_);
+  uint8_t* packet_memory = packet_buffer.MutableData();
+  // Fill the payload part of the packet memory with the pre-encoded value.
+  for (unsigned i = 0; i < 2 * payload_len_samples_; ++i)
+    packet_memory[kHeaderLenBytes + i] = encoded_sample_[i % 2];
+  WriteHeader(packet_memory);
+  // `packet` assumes ownership of `packet_memory`.
+  auto packet =
+      std::make_unique<Packet>(std::move(packet_buffer), next_arrival_time_ms_);
+  next_arrival_time_ms_ += payload_len_samples_ / samples_per_ms_;
+  return packet;
+}
+
+void ConstantPcmPacketSource::WriteHeader(uint8_t* packet_memory) {
+  packet_memory[0] = 0x80;
+  packet_memory[1] = static_cast<uint8_t>(payload_type_);
+  packet_memory[2] = seq_number_ >> 8;
+  packet_memory[3] = seq_number_ & 0xFF;
+  packet_memory[4] = timestamp_ >> 24;
+  packet_memory[5] = (timestamp_ >> 16) & 0xFF;
+  packet_memory[6] = (timestamp_ >> 8) & 0xFF;
+  packet_memory[7] = timestamp_ & 0xFF;
+  packet_memory[8] = payload_ssrc_ >> 24;
+  packet_memory[9] = (payload_ssrc_ >> 16) & 0xFF;
+  packet_memory[10] = (payload_ssrc_ >> 8) & 0xFF;
+  packet_memory[11] = payload_ssrc_ & 0xFF;
+  ++seq_number_;
+  timestamp_ += static_cast<uint32_t>(payload_len_samples_);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
new file mode 100644
index 0000000000..ab4f5c2281
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
@@ -0,0 +1,55 @@
+/*
+ *  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_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
+
+#include <stdio.h>
+
+#include <string>
+
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+
+namespace webrtc {
+namespace test {
+
+// This class implements a packet source that delivers PCM16b encoded packets
+// with a constant sample value. The payload length, constant sample value,
+// sample rate, and payload type are all set in the constructor.
+class ConstantPcmPacketSource : public PacketSource {
+ public:
+  ConstantPcmPacketSource(size_t payload_len_samples,
+                          int16_t sample_value,
+                          int sample_rate_hz,
+                          int payload_type);
+
+  ConstantPcmPacketSource(const ConstantPcmPacketSource&) = delete;
+  ConstantPcmPacketSource& operator=(const ConstantPcmPacketSource&) = delete;
+
+  std::unique_ptr<Packet> NextPacket() override;
+
+ private:
+  void WriteHeader(uint8_t* packet_memory);
+
+  const size_t kHeaderLenBytes = 12;
+  const size_t payload_len_samples_;
+  const size_t packet_len_bytes_;
+  uint8_t encoded_sample_[2];
+  const int samples_per_ms_;
+  double next_arrival_time_ms_;
+  const int payload_type_;
+  uint16_t seq_number_;
+  uint32_t timestamp_;
+  const uint32_t payload_ssrc_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
new file mode 100644
index 0000000000..87b987ddb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
@@ -0,0 +1,94 @@
+/*
+ *  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/neteq/tools/encode_neteq_input.h"
+
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace test {
+
+EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<Generator> generator,
+                                   std::unique_ptr<AudioEncoder> encoder,
+                                   int64_t input_duration_ms)
+    : generator_(std::move(generator)),
+      encoder_(std::move(encoder)),
+      input_duration_ms_(input_duration_ms) {
+  CreatePacket();
+}
+
+EncodeNetEqInput::~EncodeNetEqInput() = default;
+
+absl::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
+  RTC_DCHECK(packet_data_);
+  return static_cast<int64_t>(packet_data_->time_ms);
+}
+
+absl::optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
+  return next_output_event_ms_;
+}
+
+std::unique_ptr<NetEqInput::PacketData> EncodeNetEqInput::PopPacket() {
+  RTC_DCHECK(packet_data_);
+  // Grab the packet to return...
+  std::unique_ptr<PacketData> packet_to_return = std::move(packet_data_);
+  // ... and line up the next packet for future use.
+  CreatePacket();
+
+  return packet_to_return;
+}
+
+void EncodeNetEqInput::AdvanceOutputEvent() {
+  next_output_event_ms_ += kOutputPeriodMs;
+}
+
+bool EncodeNetEqInput::ended() const {
+  return next_output_event_ms_ > input_duration_ms_;
+}
+
+absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
+  RTC_DCHECK(packet_data_);
+  return packet_data_->header;
+}
+
+void EncodeNetEqInput::CreatePacket() {
+  // Create a new PacketData object.
+  RTC_DCHECK(!packet_data_);
+  packet_data_.reset(new NetEqInput::PacketData);
+  RTC_DCHECK_EQ(packet_data_->payload.size(), 0);
+
+  // Loop until we get a packet.
+  AudioEncoder::EncodedInfo info;
+  RTC_DCHECK(!info.send_even_if_empty);
+  int num_blocks = 0;
+  while (packet_data_->payload.size() == 0 && !info.send_even_if_empty) {
+    const size_t num_samples = rtc::CheckedDivExact(
+        static_cast<int>(encoder_->SampleRateHz() * kOutputPeriodMs), 1000);
+
+    info = encoder_->Encode(rtp_timestamp_, generator_->Generate(num_samples),
+                            &packet_data_->payload);
+
+    rtp_timestamp_ += rtc::dchecked_cast<uint32_t>(
+        num_samples * encoder_->RtpTimestampRateHz() /
+        encoder_->SampleRateHz());
+    ++num_blocks;
+  }
+  packet_data_->header.timestamp = info.encoded_timestamp;
+  packet_data_->header.payloadType = info.payload_type;
+  packet_data_->header.sequenceNumber = sequence_number_++;
+  packet_data_->time_ms = next_packet_time_ms_;
+  next_packet_time_ms_ += num_blocks * kOutputPeriodMs;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
new file mode 100644
index 0000000000..caa9ac76f4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
@@ -0,0 +1,69 @@
+/*
+ *  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_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
+
+#include <memory>
+
+#include "api/audio_codecs/audio_encoder.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+
+namespace webrtc {
+namespace test {
+
+// This class provides a NetEqInput that takes audio from a generator object and
+// encodes it using a given audio encoder.
+class EncodeNetEqInput : public NetEqInput {
+ public:
+  // Generator class, to be provided to the EncodeNetEqInput constructor.
+  class Generator {
+   public:
+    virtual ~Generator() = default;
+    // Returns the next num_samples values from the signal generator.
+    virtual rtc::ArrayView<const int16_t> Generate(size_t num_samples) = 0;
+  };
+
+  // The source will end after the given input duration.
+  EncodeNetEqInput(std::unique_ptr<Generator> generator,
+                   std::unique_ptr<AudioEncoder> encoder,
+                   int64_t input_duration_ms);
+  ~EncodeNetEqInput() override;
+
+  absl::optional<int64_t> NextPacketTime() const override;
+
+  absl::optional<int64_t> NextOutputEventTime() const override;
+
+  std::unique_ptr<PacketData> PopPacket() override;
+
+  void AdvanceOutputEvent() override;
+
+  bool ended() const override;
+
+  absl::optional<RTPHeader> NextHeader() const override;
+
+ private:
+  static constexpr int64_t kOutputPeriodMs = 10;
+
+  void CreatePacket();
+
+  std::unique_ptr<Generator> generator_;
+  std::unique_ptr<AudioEncoder> encoder_;
+  std::unique_ptr<PacketData> packet_data_;
+  uint32_t rtp_timestamp_ = 0;
+  int16_t sequence_number_ = 0;
+  int64_t next_packet_time_ms_ = 0;
+  int64_t next_output_event_ms_ = 0;
+  const int64_t input_duration_ms_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
new file mode 100644
index 0000000000..6c5e5ac2e4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
@@ -0,0 +1,169 @@
+/*
+ *  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/neteq/tools/fake_decode_from_file.h"
+
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+class FakeEncodedFrame : public AudioDecoder::EncodedAudioFrame {
+ public:
+  FakeEncodedFrame(AudioDecoder* decoder, rtc::Buffer&& payload)
+      : decoder_(decoder), payload_(std::move(payload)) {}
+
+  size_t Duration() const override {
+    const int ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    return ret < 0 ? 0 : static_cast<size_t>(ret);
+  }
+
+  absl::optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    auto speech_type = AudioDecoder::kSpeech;
+    const int ret = decoder_->Decode(
+        payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+        decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    return ret < 0 ? absl::nullopt
+                   : absl::optional<DecodeResult>(
+                         {static_cast<size_t>(ret), speech_type});
+  }
+
+  // This is to mimic OpusFrame.
+  bool IsDtxPacket() const override {
+    uint32_t original_payload_size_bytes =
+        ByteReader<uint32_t>::ReadLittleEndian(&payload_.data()[8]);
+    return original_payload_size_bytes <= 2;
+  }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+};
+
+}  // namespace
+
+std::vector<AudioDecoder::ParseResult> FakeDecodeFromFile::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new FakeEncodedFrame(this, std::move(payload)));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int FakeDecodeFromFile::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, SampleRateHz());
+
+  const int samples_to_decode = PacketDuration(encoded, encoded_len);
+  const int total_samples_to_decode = samples_to_decode * (stereo_ ? 2 : 1);
+
+  if (encoded_len == 0) {
+    // Decoder is asked to produce codec-internal comfort noise.
+    RTC_DCHECK(!encoded);  // NetEq always sends nullptr in this case.
+    RTC_DCHECK(cng_mode_);
+    RTC_DCHECK_GT(total_samples_to_decode, 0);
+    std::fill_n(decoded, total_samples_to_decode, 0);
+    *speech_type = kComfortNoise;
+    return rtc::dchecked_cast<int>(total_samples_to_decode);
+  }
+
+  RTC_CHECK_GE(encoded_len, 12);
+  uint32_t timestamp_to_decode =
+      ByteReader<uint32_t>::ReadLittleEndian(encoded);
+
+  if (next_timestamp_from_input_ &&
+      timestamp_to_decode != *next_timestamp_from_input_) {
+    // A gap in the timestamp sequence is detected. Skip the same number of
+    // samples from the file.
+    uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_;
+    RTC_CHECK(input_->Seek(jump));
+  }
+
+  next_timestamp_from_input_ = timestamp_to_decode + samples_to_decode;
+
+  uint32_t original_payload_size_bytes =
+      ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
+  if (original_payload_size_bytes <= 2) {
+    // This is a comfort noise payload.
+    RTC_DCHECK_GT(total_samples_to_decode, 0);
+    std::fill_n(decoded, total_samples_to_decode, 0);
+    *speech_type = kComfortNoise;
+    cng_mode_ = true;
+    return rtc::dchecked_cast<int>(total_samples_to_decode);
+  }
+
+  cng_mode_ = false;
+  RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded));
+
+  if (stereo_) {
+    InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2,
+                                         decoded);
+  }
+
+  *speech_type = kSpeech;
+  last_decoded_length_ = samples_to_decode;
+  return rtc::dchecked_cast<int>(total_samples_to_decode);
+}
+
+int FakeDecodeFromFile::PacketDuration(const uint8_t* encoded,
+                                       size_t encoded_len) const {
+  const uint32_t original_payload_size_bytes =
+      encoded_len < 8 + sizeof(uint32_t)
+          ? 0
+          : ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
+  const uint32_t samples_to_decode =
+      encoded_len < 4 + sizeof(uint32_t)
+          ? 0
+          : ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
+  if (encoded_len == 0) {
+    // Decoder is asked to produce codec-internal comfort noise.
+    return rtc::CheckedDivExact(SampleRateHz(), 100);
+  }
+  bool is_dtx_payload =
+      original_payload_size_bytes <= 2 || samples_to_decode == 0;
+  bool has_error_duration =
+      samples_to_decode % rtc::CheckedDivExact(SampleRateHz(), 100) != 0;
+  if (is_dtx_payload || has_error_duration) {
+    if (last_decoded_length_ > 0) {
+      // Use length of last decoded packet.
+      return rtc::dchecked_cast<int>(last_decoded_length_);
+    } else {
+      // This is the first packet to decode, and we do not know the length of
+      // it. Set it to 10 ms.
+      return rtc::CheckedDivExact(SampleRateHz(), 100);
+    }
+  }
+  return samples_to_decode;
+}
+
+void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
+                                        size_t samples,
+                                        size_t original_payload_size_bytes,
+                                        rtc::ArrayView<uint8_t> encoded) {
+  RTC_CHECK_GE(encoded.size(), 12);
+  ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp);
+  ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4],
+                                          rtc::checked_cast<uint32_t>(samples));
+  ByteWriter<uint32_t>::WriteLittleEndian(
+      &encoded[8], rtc::checked_cast<uint32_t>(original_payload_size_bytes));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
new file mode 100644
index 0000000000..7b53653998
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
@@ -0,0 +1,77 @@
+/*
+ *  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_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+
+namespace webrtc {
+namespace test {
+// Provides an AudioDecoder implementation that delivers audio data from a file.
+// The "encoded" input should contain information about what RTP timestamp the
+// encoding represents, and how many samples the decoder should produce for that
+// encoding. A helper method PrepareEncoded is provided to prepare such
+// encodings. If packets are missing, as determined from the timestamps, the
+// file reading will skip forward to match the loss.
+class FakeDecodeFromFile : public AudioDecoder {
+ public:
+  FakeDecodeFromFile(std::unique_ptr<InputAudioFile> input,
+                     int sample_rate_hz,
+                     bool stereo)
+      : input_(std::move(input)),
+        sample_rate_hz_(sample_rate_hz),
+        stereo_(stereo) {}
+
+  ~FakeDecodeFromFile() = default;
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+
+  void Reset() override {}
+
+  int SampleRateHz() const override { return sample_rate_hz_; }
+
+  size_t Channels() const override { return stereo_ ? 2 : 1; }
+
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+
+  // Helper method. Writes `timestamp`, `samples` and
+  // `original_payload_size_bytes` to `encoded` in a format that the
+  // FakeDecodeFromFile decoder will understand. `encoded` must be at least 12
+  // bytes long.
+  static void PrepareEncoded(uint32_t timestamp,
+                             size_t samples,
+                             size_t original_payload_size_bytes,
+                             rtc::ArrayView<uint8_t> encoded);
+
+ private:
+  std::unique_ptr<InputAudioFile> input_;
+  absl::optional<uint32_t> next_timestamp_from_input_;
+  const int sample_rate_hz_;
+  const bool stereo_;
+  size_t last_decoded_length_ = 0;
+  bool cng_mode_ = false;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc
new file mode 100644
index 0000000000..3c33aabf1c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.cc
@@ -0,0 +1,79 @@
+/*
+ *  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/neteq/tools/initial_packet_inserter_neteq_input.h"
+
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+InitialPacketInserterNetEqInput::InitialPacketInserterNetEqInput(
+    std::unique_ptr<NetEqInput> source,
+    int number_of_initial_packets,
+    int sample_rate_hz)
+    : source_(std::move(source)),
+      packets_to_insert_(number_of_initial_packets),
+      sample_rate_hz_(sample_rate_hz) {}
+
+absl::optional<int64_t> InitialPacketInserterNetEqInput::NextPacketTime()
+    const {
+  return source_->NextPacketTime();
+}
+
+absl::optional<int64_t> InitialPacketInserterNetEqInput::NextOutputEventTime()
+    const {
+  return source_->NextOutputEventTime();
+}
+
+std::unique_ptr<InitialPacketInserterNetEqInput::PacketData>
+InitialPacketInserterNetEqInput::PopPacket() {
+  if (!first_packet_) {
+    first_packet_ = source_->PopPacket();
+    if (!first_packet_) {
+      // The source has no packets, so we should not insert any dummy packets.
+      packets_to_insert_ = 0;
+    }
+  }
+  if (packets_to_insert_ > 0) {
+    RTC_CHECK(first_packet_);
+    auto dummy_packet = std::unique_ptr<PacketData>(new PacketData());
+    dummy_packet->header = first_packet_->header;
+    dummy_packet->payload = rtc::Buffer(first_packet_->payload.data(),
+                                        first_packet_->payload.size());
+    dummy_packet->time_ms = first_packet_->time_ms;
+    dummy_packet->header.sequenceNumber -= packets_to_insert_;
+    // This assumes 20ms per packet.
+    dummy_packet->header.timestamp -=
+        20 * sample_rate_hz_ * packets_to_insert_ / 1000;
+    packets_to_insert_--;
+    return dummy_packet;
+  }
+  return source_->PopPacket();
+}
+
+void InitialPacketInserterNetEqInput::AdvanceOutputEvent() {
+  source_->AdvanceOutputEvent();
+}
+
+bool InitialPacketInserterNetEqInput::ended() const {
+  return source_->ended();
+}
+
+absl::optional<RTPHeader> InitialPacketInserterNetEqInput::NextHeader() const {
+  return source_->NextHeader();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h
new file mode 100644
index 0000000000..bd20a7aecf
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h
@@ -0,0 +1,46 @@
+/*
+ *  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_NETEQ_TOOLS_INITIAL_PACKET_INSERTER_NETEQ_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_INITIAL_PACKET_INSERTER_NETEQ_INPUT_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+
+namespace webrtc {
+namespace test {
+
+// Wrapper class that can insert a number of packets at the start of the
+// simulation.
+class InitialPacketInserterNetEqInput final : public NetEqInput {
+ public:
+  InitialPacketInserterNetEqInput(std::unique_ptr<NetEqInput> source,
+                                  int number_of_initial_packets,
+                                  int sample_rate_hz);
+  absl::optional<int64_t> NextPacketTime() const override;
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  std::unique_ptr<PacketData> PopPacket() override;
+  void AdvanceOutputEvent() override;
+  bool ended() const override;
+  absl::optional<RTPHeader> NextHeader() const override;
+
+ private:
+  const std::unique_ptr<NetEqInput> source_;
+  int packets_to_insert_;
+  const int sample_rate_hz_;
+  std::unique_ptr<PacketData> first_packet_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_INITIAL_PACKET_INSERTER_NETEQ_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
new file mode 100644
index 0000000000..b077dbff21
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
@@ -0,0 +1,96 @@
+/*
+ *  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 "modules/audio_coding/neteq/tools/input_audio_file.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+InputAudioFile::InputAudioFile(absl::string_view file_name, bool loop_at_end)
+    : loop_at_end_(loop_at_end) {
+  fp_ = fopen(std::string(file_name).c_str(), "rb");
+  RTC_DCHECK(fp_) << file_name << " could not be opened.";
+}
+
+InputAudioFile::~InputAudioFile() {
+  RTC_DCHECK(fp_);
+  fclose(fp_);
+}
+
+bool InputAudioFile::Read(size_t samples, int16_t* destination) {
+  if (!fp_) {
+    return false;
+  }
+  size_t samples_read = fread(destination, sizeof(int16_t), samples, fp_);
+  if (samples_read < samples) {
+    if (!loop_at_end_) {
+      return false;
+    }
+    // Rewind and read the missing samples.
+    rewind(fp_);
+    size_t missing_samples = samples - samples_read;
+    if (fread(destination + samples_read, sizeof(int16_t), missing_samples,
+              fp_) < missing_samples) {
+      // Could not read enough even after rewinding the file.
+      return false;
+    }
+  }
+  return true;
+}
+
+bool InputAudioFile::Seek(int samples) {
+  if (!fp_) {
+    return false;
+  }
+  // Find file boundaries.
+  const long current_pos = ftell(fp_);
+  RTC_CHECK_NE(EOF, current_pos)
+      << "Error returned when getting file position.";
+  RTC_CHECK_EQ(0, fseek(fp_, 0, SEEK_END));  // Move to end of file.
+  const long file_size = ftell(fp_);
+  RTC_CHECK_NE(EOF, file_size) << "Error returned when getting file position.";
+  // Find new position.
+  long new_pos = current_pos + sizeof(int16_t) * samples;  // Samples to bytes.
+  if (loop_at_end_) {
+    new_pos = new_pos % file_size;  // Wrap around the end of the file.
+    if (new_pos < 0) {
+      // For negative values of new_pos, newpos % file_size will also be
+      // negative. To get the correct result it's needed to add file_size.
+      new_pos += file_size;
+    }
+  } else {
+    new_pos = new_pos > file_size ? file_size : new_pos;  // Don't loop.
+  }
+  RTC_CHECK_GE(new_pos, 0)
+      << "Trying to move to before the beginning of the file";
+  // Move to new position relative to the beginning of the file.
+  RTC_CHECK_EQ(0, fseek(fp_, new_pos, SEEK_SET));
+  return true;
+}
+
+void InputAudioFile::DuplicateInterleaved(const int16_t* source,
+                                          size_t samples,
+                                          size_t channels,
+                                          int16_t* destination) {
+  // Start from the end of `source` and `destination`, and work towards the
+  // beginning. This is to allow in-place interleaving of the same array (i.e.,
+  // `source` and `destination` are the same array).
+  for (int i = static_cast<int>(samples - 1); i >= 0; --i) {
+    for (int j = static_cast<int>(channels - 1); j >= 0; --j) {
+      destination[i * channels + j] = source[i];
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.h
new file mode 100644
index 0000000000..f538b295a3
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file.h
@@ -0,0 +1,62 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
+
+#include <stdio.h>
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for handling a looping input audio file.
+class InputAudioFile {
+ public:
+  explicit InputAudioFile(absl::string_view file_name, bool loop_at_end = true);
+
+  virtual ~InputAudioFile();
+
+  InputAudioFile(const InputAudioFile&) = delete;
+  InputAudioFile& operator=(const InputAudioFile&) = delete;
+
+  // Reads `samples` elements from source file to `destination`. Returns true
+  // if the read was successful, otherwise false. If the file end is reached,
+  // the file is rewound and reading continues from the beginning.
+  // The output `destination` must have the capacity to hold `samples` elements.
+  virtual bool Read(size_t samples, int16_t* destination);
+
+  // Fast-forwards (`samples` > 0) or -backwards (`samples` < 0) the file by the
+  // indicated number of samples. Just like Read(), Seek() starts over at the
+  // beginning of the file if the end is reached. However, seeking backwards
+  // past the beginning of the file is not possible.
+  virtual bool Seek(int samples);
+
+  // Creates a multi-channel signal from a mono signal. Each sample is repeated
+  // `channels` times to create an interleaved multi-channel signal where all
+  // channels are identical. The output `destination` must have the capacity to
+  // hold samples * channels elements. Note that `source` and `destination` can
+  // be the same array (i.e., point to the same address).
+  static void DuplicateInterleaved(const int16_t* source,
+                                   size_t samples,
+                                   size_t channels,
+                                   int16_t* destination);
+
+ private:
+  FILE* fp_;
+  const bool loop_at_end_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
new file mode 100644
index 0000000000..52f7ea82a0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
@@ -0,0 +1,59 @@
+/*
+ *  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.
+ */
+
+// Unit tests for test InputAudioFile class.
+
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+TEST(TestInputAudioFile, DuplicateInterleaveSeparateSrcDst) {
+  static const size_t kSamples = 10;
+  static const size_t kChannels = 2;
+  int16_t input[kSamples];
+  for (size_t i = 0; i < kSamples; ++i) {
+    input[i] = rtc::checked_cast<int16_t>(i);
+  }
+  int16_t output[kSamples * kChannels];
+  InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, output);
+
+  // Verify output
+  int16_t* output_ptr = output;
+  for (size_t i = 0; i < kSamples; ++i) {
+    for (size_t j = 0; j < kChannels; ++j) {
+      EXPECT_EQ(static_cast<int16_t>(i), *output_ptr++);
+    }
+  }
+}
+
+TEST(TestInputAudioFile, DuplicateInterleaveSameSrcDst) {
+  static const size_t kSamples = 10;
+  static const size_t kChannels = 5;
+  int16_t input[kSamples * kChannels];
+  for (size_t i = 0; i < kSamples; ++i) {
+    input[i] = rtc::checked_cast<int16_t>(i);
+  }
+  InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, input);
+
+  // Verify output
+  int16_t* output_ptr = input;
+  for (size_t i = 0; i < kSamples; ++i) {
+    for (size_t j = 0; j < kChannels; ++j) {
+      EXPECT_EQ(static_cast<int16_t>(i), *output_ptr++);
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
new file mode 100644
index 0000000000..9e77457775
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
@@ -0,0 +1,307 @@
+/*
+ *  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/neteq/tools/neteq_delay_analyzer.h"
+
+#include <algorithm>
+#include <fstream>
+#include <ios>
+#include <iterator>
+#include <limits>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+constexpr char kArrivalDelayX[] = "arrival_delay_x";
+constexpr char kArrivalDelayY[] = "arrival_delay_y";
+constexpr char kTargetDelayX[] = "target_delay_x";
+constexpr char kTargetDelayY[] = "target_delay_y";
+constexpr char kPlayoutDelayX[] = "playout_delay_x";
+constexpr char kPlayoutDelayY[] = "playout_delay_y";
+
+// Helper function for NetEqDelayAnalyzer::CreateGraphs. Returns the
+// interpolated value of a function at the point x. Vector x_vec contains the
+// sample points, and y_vec contains the function values at these points. The
+// return value is a linear interpolation between y_vec values.
+double LinearInterpolate(double x,
+                         const std::vector<int64_t>& x_vec,
+                         const std::vector<int64_t>& y_vec) {
+  // Find first element which is larger than x.
+  auto it = std::upper_bound(x_vec.begin(), x_vec.end(), x);
+  if (it == x_vec.end()) {
+    --it;
+  }
+  const size_t upper_ix = it - x_vec.begin();
+
+  size_t lower_ix;
+  if (upper_ix == 0 || x_vec[upper_ix] <= x) {
+    lower_ix = upper_ix;
+  } else {
+    lower_ix = upper_ix - 1;
+  }
+  double y;
+  if (lower_ix == upper_ix) {
+    y = y_vec[lower_ix];
+  } else {
+    RTC_DCHECK_NE(x_vec[lower_ix], x_vec[upper_ix]);
+    y = (x - x_vec[lower_ix]) * (y_vec[upper_ix] - y_vec[lower_ix]) /
+            (x_vec[upper_ix] - x_vec[lower_ix]) +
+        y_vec[lower_ix];
+  }
+  return y;
+}
+
+void PrintDelays(const NetEqDelayAnalyzer::Delays& delays,
+                 int64_t ref_time_ms,
+                 absl::string_view var_name_x,
+                 absl::string_view var_name_y,
+                 std::ofstream& output,
+                 absl::string_view terminator = "") {
+  output << var_name_x << " = [ ";
+  for (const std::pair<int64_t, float>& delay : delays) {
+    output << (delay.first - ref_time_ms) / 1000.f << ", ";
+  }
+  output << "]" << terminator << std::endl;
+
+  output << var_name_y << " = [ ";
+  for (const std::pair<int64_t, float>& delay : delays) {
+    output << delay.second << ", ";
+  }
+  output << "]" << terminator << std::endl;
+}
+
+}  // namespace
+
+void NetEqDelayAnalyzer::AfterInsertPacket(
+    const test::NetEqInput::PacketData& packet,
+    NetEq* neteq) {
+  data_.insert(
+      std::make_pair(packet.header.timestamp, TimingData(packet.time_ms)));
+  ssrcs_.insert(packet.header.ssrc);
+  payload_types_.insert(packet.header.payloadType);
+}
+
+void NetEqDelayAnalyzer::BeforeGetAudio(NetEq* neteq) {
+  last_sync_buffer_ms_ = neteq->SyncBufferSizeMs();
+}
+
+void NetEqDelayAnalyzer::AfterGetAudio(int64_t time_now_ms,
+                                       const AudioFrame& audio_frame,
+                                       bool /*muted*/,
+                                       NetEq* neteq) {
+  get_audio_time_ms_.push_back(time_now_ms);
+  for (const RtpPacketInfo& info : audio_frame.packet_infos_) {
+    auto it = data_.find(info.rtp_timestamp());
+    if (it == data_.end()) {
+      // This is a packet that was split out from another packet. Skip it.
+      continue;
+    }
+    auto& it_timing = it->second;
+    RTC_CHECK(!it_timing.decode_get_audio_count)
+        << "Decode time already written";
+    it_timing.decode_get_audio_count = get_audio_count_;
+    RTC_CHECK(!it_timing.sync_delay_ms) << "Decode time already written";
+    it_timing.sync_delay_ms = last_sync_buffer_ms_;
+    it_timing.target_delay_ms = neteq->TargetDelayMs();
+    it_timing.current_delay_ms = neteq->FilteredCurrentDelayMs();
+  }
+  last_sample_rate_hz_ = audio_frame.sample_rate_hz_;
+  ++get_audio_count_;
+}
+
+void NetEqDelayAnalyzer::CreateGraphs(Delays* arrival_delay_ms,
+                                      Delays* corrected_arrival_delay_ms,
+                                      Delays* playout_delay_ms,
+                                      Delays* target_delay_ms) const {
+  if (get_audio_time_ms_.empty()) {
+    return;
+  }
+  // Create nominal_get_audio_time_ms, a vector starting at
+  // get_audio_time_ms_[0] and increasing by 10 for each element.
+  std::vector<int64_t> nominal_get_audio_time_ms(get_audio_time_ms_.size());
+  nominal_get_audio_time_ms[0] = get_audio_time_ms_[0];
+  std::transform(
+      nominal_get_audio_time_ms.begin(), nominal_get_audio_time_ms.end() - 1,
+      nominal_get_audio_time_ms.begin() + 1, [](int64_t& x) { return x + 10; });
+  RTC_DCHECK(
+      std::is_sorted(get_audio_time_ms_.begin(), get_audio_time_ms_.end()));
+
+  std::vector<double> rtp_timestamps_ms;
+  double offset = std::numeric_limits<double>::max();
+  RtpTimestampUnwrapper unwrapper;
+  // This loop traverses data_ and populates rtp_timestamps_ms as well as
+  // calculates the base offset.
+  for (auto& d : data_) {
+    rtp_timestamps_ms.push_back(
+        static_cast<double>(unwrapper.Unwrap(d.first)) /
+        rtc::CheckedDivExact(last_sample_rate_hz_, 1000));
+    offset =
+        std::min(offset, d.second.arrival_time_ms - rtp_timestamps_ms.back());
+  }
+
+  // This loop traverses the data again and populates the graph vectors. The
+  // reason to have two loops and traverse twice is that the offset cannot be
+  // known until the first traversal is done. Meanwhile, the final offset must
+  // be known already at the start of this second loop.
+  size_t i = 0;
+  for (const auto& data : data_) {
+    const double offset_send_time_ms = rtp_timestamps_ms[i++] + offset;
+    const auto& timing = data.second;
+    corrected_arrival_delay_ms->push_back(std::make_pair(
+        timing.arrival_time_ms,
+        LinearInterpolate(timing.arrival_time_ms, get_audio_time_ms_,
+                          nominal_get_audio_time_ms) -
+            offset_send_time_ms));
+    arrival_delay_ms->push_back(std::make_pair(
+        timing.arrival_time_ms, timing.arrival_time_ms - offset_send_time_ms));
+
+    if (timing.decode_get_audio_count) {
+      // This packet was decoded.
+      RTC_DCHECK(timing.sync_delay_ms);
+      const int64_t get_audio_time =
+          *timing.decode_get_audio_count * 10 + get_audio_time_ms_[0];
+      const float playout_ms =
+          get_audio_time + *timing.sync_delay_ms - offset_send_time_ms;
+      playout_delay_ms->push_back(std::make_pair(get_audio_time, playout_ms));
+      RTC_DCHECK(timing.target_delay_ms);
+      RTC_DCHECK(timing.current_delay_ms);
+      const float target =
+          playout_ms - *timing.current_delay_ms + *timing.target_delay_ms;
+      target_delay_ms->push_back(std::make_pair(get_audio_time, target));
+    }
+  }
+}
+
+void NetEqDelayAnalyzer::CreateMatlabScript(
+    absl::string_view script_name) const {
+  Delays arrival_delay_ms;
+  Delays corrected_arrival_delay_ms;
+  Delays playout_delay_ms;
+  Delays target_delay_ms;
+  CreateGraphs(&arrival_delay_ms, &corrected_arrival_delay_ms,
+               &playout_delay_ms, &target_delay_ms);
+
+  // Maybe better to find the actually smallest timestamp, to surely avoid
+  // x-axis starting from negative.
+  const int64_t ref_time_ms = arrival_delay_ms.front().first;
+
+  // Create an output file stream to Matlab script file.
+  std::ofstream output(std::string{script_name});
+
+  PrintDelays(corrected_arrival_delay_ms, ref_time_ms, kArrivalDelayX,
+              kArrivalDelayY, output, ";");
+
+  // PrintDelays(corrected_arrival_delay_x, kCorrectedArrivalDelayX,
+  // kCorrectedArrivalDelayY, output);
+
+  PrintDelays(playout_delay_ms, ref_time_ms, kPlayoutDelayX, kPlayoutDelayY,
+              output, ";");
+
+  PrintDelays(target_delay_ms, ref_time_ms, kTargetDelayX, kTargetDelayY,
+              output, ";");
+
+  output << "h=plot(" << kArrivalDelayX << ", " << kArrivalDelayY << ", "
+         << kTargetDelayX << ", " << kTargetDelayY << ", 'g.', "
+         << kPlayoutDelayX << ", " << kPlayoutDelayY << ");" << std::endl;
+  output << "set(h(1),'color',0.75*[1 1 1]);" << std::endl;
+  output << "set(h(2),'markersize',6);" << std::endl;
+  output << "set(h(3),'linew',1.5);" << std::endl;
+  output << "ax1=axis;" << std::endl;
+  output << "axis tight" << std::endl;
+  output << "ax2=axis;" << std::endl;
+  output << "axis([ax2(1:3) ax1(4)])" << std::endl;
+  output << "xlabel('time [s]');" << std::endl;
+  output << "ylabel('relative delay [ms]');" << std::endl;
+  if (!ssrcs_.empty()) {
+    auto ssrc_it = ssrcs_.cbegin();
+    output << "title('SSRC: 0x" << std::hex << static_cast<int64_t>(*ssrc_it++);
+    while (ssrc_it != ssrcs_.end()) {
+      output << ", 0x" << std::hex << static_cast<int64_t>(*ssrc_it++);
+    }
+    output << std::dec;
+    auto pt_it = payload_types_.cbegin();
+    output << "; Payload Types: " << *pt_it++;
+    while (pt_it != payload_types_.end()) {
+      output << ", " << *pt_it++;
+    }
+    output << "');" << std::endl;
+  }
+}
+
+void NetEqDelayAnalyzer::CreatePythonScript(
+    absl::string_view script_name) const {
+  Delays arrival_delay_ms;
+  Delays corrected_arrival_delay_ms;
+  Delays playout_delay_ms;
+  Delays target_delay_ms;
+  CreateGraphs(&arrival_delay_ms, &corrected_arrival_delay_ms,
+               &playout_delay_ms, &target_delay_ms);
+
+  // Maybe better to find the actually smallest timestamp, to surely avoid
+  // x-axis starting from negative.
+  const int64_t ref_time_ms = arrival_delay_ms.front().first;
+
+  // Create an output file stream to the python script file.
+  std::ofstream output(std::string{script_name});
+
+  // Necessary includes
+  output << "import numpy as np" << std::endl;
+  output << "import matplotlib.pyplot as plt" << std::endl;
+
+  PrintDelays(corrected_arrival_delay_ms, ref_time_ms, kArrivalDelayX,
+              kArrivalDelayY, output);
+
+  // PrintDelays(corrected_arrival_delay_x, kCorrectedArrivalDelayX,
+  // kCorrectedArrivalDelayY, output);
+
+  PrintDelays(playout_delay_ms, ref_time_ms, kPlayoutDelayX, kPlayoutDelayY,
+              output);
+
+  PrintDelays(target_delay_ms, ref_time_ms, kTargetDelayX, kTargetDelayY,
+              output);
+
+  output << "if __name__ == '__main__':" << std::endl;
+  output << "  h=plt.plot(" << kArrivalDelayX << ", " << kArrivalDelayY << ", "
+         << kTargetDelayX << ", " << kTargetDelayY << ", 'g.', "
+         << kPlayoutDelayX << ", " << kPlayoutDelayY << ")" << std::endl;
+  output << "  plt.setp(h[0],'color',[.75, .75, .75])" << std::endl;
+  output << "  plt.setp(h[1],'markersize',6)" << std::endl;
+  output << "  plt.setp(h[2],'linewidth',1.5)" << std::endl;
+  output << "  plt.axis('tight')" << std::endl;
+  output << "  plt.xlabel('time [s]')" << std::endl;
+  output << "  plt.ylabel('relative delay [ms]')" << std::endl;
+  if (!ssrcs_.empty()) {
+    auto ssrc_it = ssrcs_.cbegin();
+    output << "  plt.legend((\"arrival delay\", \"target delay\", \"playout "
+              "delay\"))"
+           << std::endl;
+    output << "  plt.title('SSRC: 0x" << std::hex
+           << static_cast<int64_t>(*ssrc_it++);
+    while (ssrc_it != ssrcs_.end()) {
+      output << ", 0x" << std::hex << static_cast<int64_t>(*ssrc_it++);
+    }
+    output << std::dec;
+    auto pt_it = payload_types_.cbegin();
+    output << "; Payload Types: " << *pt_it++;
+    while (pt_it != payload_types_.end()) {
+      output << ", " << *pt_it++;
+    }
+    output << "')" << std::endl;
+  }
+  output << "  plt.show()" << std::endl;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
new file mode 100644
index 0000000000..ffcba5843f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
+
+#include <map>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqDelayAnalyzer : public test::NetEqPostInsertPacket,
+                           public test::NetEqGetAudioCallback {
+ public:
+  void AfterInsertPacket(const test::NetEqInput::PacketData& packet,
+                         NetEq* neteq) override;
+
+  void BeforeGetAudio(NetEq* neteq) override;
+
+  void AfterGetAudio(int64_t time_now_ms,
+                     const AudioFrame& audio_frame,
+                     bool muted,
+                     NetEq* neteq) override;
+
+  using Delays = std::vector<std::pair<int64_t, float>>;
+  void CreateGraphs(Delays* arrival_delay_ms,
+                    Delays* corrected_arrival_delay_ms,
+                    Delays* playout_delay_ms,
+                    Delays* target_delay_ms) const;
+
+  // Creates a matlab script with file name script_name. When executed in
+  // Matlab, the script will generate graphs with the same timing information
+  // as provided by CreateGraphs.
+  void CreateMatlabScript(absl::string_view script_name) const;
+
+  // Creates a python script with file name `script_name`. When executed in
+  // Python, the script will generate graphs with the same timing information
+  // as provided by CreateGraphs.
+  void CreatePythonScript(absl::string_view script_name) const;
+
+ private:
+  struct TimingData {
+    explicit TimingData(int64_t at) : arrival_time_ms(at) {}
+    int64_t arrival_time_ms;
+    absl::optional<int64_t> decode_get_audio_count;
+    absl::optional<int64_t> sync_delay_ms;
+    absl::optional<int> target_delay_ms;
+    absl::optional<int> current_delay_ms;
+  };
+  std::map<uint32_t, TimingData> data_;
+  std::vector<int64_t> get_audio_time_ms_;
+  size_t get_audio_count_ = 0;
+  size_t last_sync_buffer_ms_ = 0;
+  int last_sample_rate_hz_ = 0;
+  std::set<uint32_t> ssrcs_;
+  std::set<int> payload_types_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
new file mode 100644
index 0000000000..0c1f27799a
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
@@ -0,0 +1,68 @@
+/*
+ *  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/neteq/tools/neteq_event_log_input.h"
+
+#include <limits>
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/rtc_event_log_source.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqEventLogInput* NetEqEventLogInput::CreateFromFile(
+    absl::string_view file_name,
+    absl::optional<uint32_t> ssrc_filter) {
+  auto event_log_src =
+      RtcEventLogSource::CreateFromFile(file_name, ssrc_filter);
+  if (!event_log_src) {
+    return nullptr;
+  }
+  return new NetEqEventLogInput(std::move(event_log_src));
+}
+
+NetEqEventLogInput* NetEqEventLogInput::CreateFromString(
+    absl::string_view file_contents,
+    absl::optional<uint32_t> ssrc_filter) {
+  auto event_log_src =
+      RtcEventLogSource::CreateFromString(file_contents, ssrc_filter);
+  if (!event_log_src) {
+    return nullptr;
+  }
+  return new NetEqEventLogInput(std::move(event_log_src));
+}
+
+absl::optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
+  return next_output_event_ms_;
+}
+
+void NetEqEventLogInput::AdvanceOutputEvent() {
+  next_output_event_ms_ = source_->NextAudioOutputEventMs();
+  if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) {
+    next_output_event_ms_ = absl::nullopt;
+  }
+}
+
+PacketSource* NetEqEventLogInput::source() {
+  return source_.get();
+}
+
+NetEqEventLogInput::NetEqEventLogInput(
+    std::unique_ptr<RtcEventLogSource> source)
+    : source_(std::move(source)) {
+  LoadNextPacket();
+  AdvanceOutputEvent();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h
new file mode 100644
index 0000000000..c947ee1fc0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h
@@ -0,0 +1,51 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+namespace test {
+
+class RtcEventLogSource;
+
+// Implementation of NetEqPacketSourceInput to be used with an
+// RtcEventLogSource.
+class NetEqEventLogInput final : public NetEqPacketSourceInput {
+ public:
+  static NetEqEventLogInput* CreateFromFile(
+      absl::string_view file_name,
+      absl::optional<uint32_t> ssrc_filter);
+  static NetEqEventLogInput* CreateFromString(
+      absl::string_view file_contents,
+      absl::optional<uint32_t> ssrc_filter);
+
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  void AdvanceOutputEvent() override;
+
+ protected:
+  PacketSource* source() override;
+
+ private:
+  NetEqEventLogInput(std::unique_ptr<RtcEventLogSource> source);
+  std::unique_ptr<RtcEventLogSource> source_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.cc
new file mode 100644
index 0000000000..de416348f1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.cc
@@ -0,0 +1,93 @@
+/*
+ *  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/neteq/tools/neteq_input.h"
+
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqInput::PacketData::PacketData() = default;
+NetEqInput::PacketData::~PacketData() = default;
+
+std::string NetEqInput::PacketData::ToString() const {
+  rtc::StringBuilder ss;
+  ss << "{"
+        "time_ms: "
+     << static_cast<int64_t>(time_ms)
+     << ", "
+        "header: {"
+        "pt: "
+     << static_cast<int>(header.payloadType)
+     << ", "
+        "sn: "
+     << header.sequenceNumber
+     << ", "
+        "ts: "
+     << header.timestamp
+     << ", "
+        "ssrc: "
+     << header.ssrc
+     << "}, "
+        "payload bytes: "
+     << payload.size() << "}";
+  return ss.Release();
+}
+
+TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
+                                             int64_t duration_ms)
+    : input_(std::move(input)),
+      start_time_ms_(input_->NextEventTime()),
+      duration_ms_(duration_ms) {}
+
+TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default;
+
+absl::optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
+  return ended_ ? absl::nullopt : input_->NextPacketTime();
+}
+
+absl::optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
+  return ended_ ? absl::nullopt : input_->NextOutputEventTime();
+}
+
+std::unique_ptr<NetEqInput::PacketData> TimeLimitedNetEqInput::PopPacket() {
+  if (ended_) {
+    return std::unique_ptr<PacketData>();
+  }
+  auto packet = input_->PopPacket();
+  MaybeSetEnded();
+  return packet;
+}
+
+void TimeLimitedNetEqInput::AdvanceOutputEvent() {
+  if (!ended_) {
+    input_->AdvanceOutputEvent();
+    MaybeSetEnded();
+  }
+}
+
+bool TimeLimitedNetEqInput::ended() const {
+  return ended_ || input_->ended();
+}
+
+absl::optional<RTPHeader> TimeLimitedNetEqInput::NextHeader() const {
+  return ended_ ? absl::nullopt : input_->NextHeader();
+}
+
+void TimeLimitedNetEqInput::MaybeSetEnded() {
+  if (NextEventTime() && start_time_ms_ &&
+      *NextEventTime() - *start_time_ms_ > duration_ms_) {
+    ended_ = true;
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.h
new file mode 100644
index 0000000000..3a66264043
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_input.h
@@ -0,0 +1,107 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
+
+#include <algorithm>
+#include <memory>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for input to the NetEqTest class.
+class NetEqInput {
+ public:
+  struct PacketData {
+    PacketData();
+    ~PacketData();
+    std::string ToString() const;
+
+    RTPHeader header;
+    rtc::Buffer payload;
+    int64_t time_ms;
+  };
+
+  virtual ~NetEqInput() = default;
+
+  // Returns at what time (in ms) NetEq::InsertPacket should be called next, or
+  // empty if the source is out of packets.
+  virtual absl::optional<int64_t> NextPacketTime() const = 0;
+
+  // Returns at what time (in ms) NetEq::GetAudio should be called next, or
+  // empty if no more output events are available.
+  virtual absl::optional<int64_t> NextOutputEventTime() const = 0;
+
+  // Returns the time (in ms) for the next event from either NextPacketTime()
+  // or NextOutputEventTime(), or empty if both are out of events.
+  absl::optional<int64_t> NextEventTime() const {
+    const auto a = NextPacketTime();
+    const auto b = NextOutputEventTime();
+    // Return the minimum of non-empty `a` and `b`, or empty if both are empty.
+    if (a) {
+      return b ? std::min(*a, *b) : a;
+    }
+    return b ? b : absl::nullopt;
+  }
+
+  // Returns the next packet to be inserted into NetEq. The packet following the
+  // returned one is pre-fetched in the NetEqInput object, such that future
+  // calls to NextPacketTime() or NextHeader() will return information from that
+  // packet.
+  virtual std::unique_ptr<PacketData> PopPacket() = 0;
+
+  // Move to the next output event. This will make NextOutputEventTime() return
+  // a new value (potentially the same if several output events share the same
+  // time).
+  virtual void AdvanceOutputEvent() = 0;
+
+  // Returns true if the source has come to an end. An implementation must
+  // eventually return true from this method, or the test will end up in an
+  // infinite loop.
+  virtual bool ended() const = 0;
+
+  // Returns the RTP header for the next packet, i.e., the packet that will be
+  // delivered next by PopPacket().
+  virtual absl::optional<RTPHeader> NextHeader() const = 0;
+};
+
+// Wrapper class to impose a time limit on a NetEqInput object, typically
+// another time limit than what the object itself provides. For example, an
+// input taken from a file can be cut shorter by wrapping it in this class.
+class TimeLimitedNetEqInput : public NetEqInput {
+ public:
+  TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
+  ~TimeLimitedNetEqInput() override;
+  absl::optional<int64_t> NextPacketTime() const override;
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  std::unique_ptr<PacketData> PopPacket() override;
+  void AdvanceOutputEvent() override;
+  bool ended() const override;
+  absl::optional<RTPHeader> NextHeader() const override;
+
+ private:
+  void MaybeSetEnded();
+
+  std::unique_ptr<NetEqInput> input_;
+  const absl::optional<int64_t> start_time_ms_;
+  const int64_t duration_ms_;
+  bool ended_ = false;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
new file mode 100644
index 0000000000..55a5653238
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
@@ -0,0 +1,90 @@
+/*
+ *  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/neteq/tools/neteq_packet_source_input.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {}
+
+absl::optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
+  return packet_
+             ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
+             : absl::nullopt;
+}
+
+absl::optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
+  return packet_ ? absl::optional<RTPHeader>(packet_->header()) : absl::nullopt;
+}
+
+void NetEqPacketSourceInput::LoadNextPacket() {
+  packet_ = source()->NextPacket();
+}
+
+std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() {
+  if (!packet_) {
+    return std::unique_ptr<PacketData>();
+  }
+  std::unique_ptr<PacketData> packet_data(new PacketData);
+  packet_data->header = packet_->header();
+  if (packet_->payload_length_bytes() == 0 &&
+      packet_->virtual_payload_length_bytes() > 0) {
+    // This is a header-only "dummy" packet. Set the payload to all zeros, with
+    // length according to the virtual length.
+    packet_data->payload.SetSize(packet_->virtual_payload_length_bytes());
+    std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
+  } else {
+    packet_data->payload.SetData(packet_->payload(),
+                                 packet_->payload_length_bytes());
+  }
+  packet_data->time_ms = packet_->time_ms();
+
+  LoadNextPacket();
+
+  return packet_data;
+}
+
+NetEqRtpDumpInput::NetEqRtpDumpInput(absl::string_view file_name,
+                                     const RtpHeaderExtensionMap& hdr_ext_map,
+                                     absl::optional<uint32_t> ssrc_filter)
+    : source_(RtpFileSource::Create(file_name, ssrc_filter)) {
+  for (const auto& ext_pair : hdr_ext_map) {
+    source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
+  }
+  LoadNextPacket();
+}
+
+absl::optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
+  return next_output_event_ms_;
+}
+
+void NetEqRtpDumpInput::AdvanceOutputEvent() {
+  if (next_output_event_ms_) {
+    *next_output_event_ms_ += kOutputPeriodMs;
+  }
+  if (!NextPacketTime()) {
+    next_output_event_ms_ = absl::nullopt;
+  }
+}
+
+PacketSource* NetEqRtpDumpInput::source() {
+  return source_.get();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
new file mode 100644
index 0000000000..407fa491b1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
@@ -0,0 +1,70 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+namespace test {
+
+class RtpFileSource;
+
+// An adapter class to dress up a PacketSource object as a NetEqInput.
+class NetEqPacketSourceInput : public NetEqInput {
+ public:
+  using RtpHeaderExtensionMap = std::map<int, webrtc::RTPExtensionType>;
+
+  NetEqPacketSourceInput();
+  absl::optional<int64_t> NextPacketTime() const override;
+  std::unique_ptr<PacketData> PopPacket() override;
+  absl::optional<RTPHeader> NextHeader() const override;
+  bool ended() const override { return !next_output_event_ms_; }
+
+ protected:
+  virtual PacketSource* source() = 0;
+  void LoadNextPacket();
+
+  absl::optional<int64_t> next_output_event_ms_;
+
+ private:
+  std::unique_ptr<Packet> packet_;
+};
+
+// Implementation of NetEqPacketSourceInput to be used with an RtpFileSource.
+class NetEqRtpDumpInput final : public NetEqPacketSourceInput {
+ public:
+  NetEqRtpDumpInput(absl::string_view file_name,
+                    const RtpHeaderExtensionMap& hdr_ext_map,
+                    absl::optional<uint32_t> ssrc_filter);
+
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  void AdvanceOutputEvent() override;
+
+ protected:
+  PacketSource* source() override;
+
+ private:
+  static constexpr int64_t kOutputPeriodMs = 10;
+
+  std::unique_ptr<RtpFileSource> source_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
new file mode 100644
index 0000000000..ccaa87b5e8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -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.
+ */
+
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
+
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/clock.h"
+#include "test/testsupport/file_utils.h"
+
+using webrtc::NetEq;
+using webrtc::test::AudioLoop;
+using webrtc::test::RtpGenerator;
+
+namespace webrtc {
+namespace test {
+
+int64_t NetEqPerformanceTest::Run(int runtime_ms,
+                                  int lossrate,
+                                  double drift_factor) {
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  const int kSampRateHz = 32000;
+  const std::string kDecoderName = "pcm16-swb32";
+  const int kPayloadType = 95;
+
+  // Initialize NetEq instance.
+  NetEq::Config config;
+  config.sample_rate_hz = kSampRateHz;
+  webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
+  auto audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
+  auto neteq =
+      DefaultNetEqFactory().CreateNetEq(config, audio_decoder_factory, clock);
+  // Register decoder in `neteq`.
+  if (!neteq->RegisterPayloadType(kPayloadType,
+                                  SdpAudioFormat("l16", kSampRateHz, 1)))
+    return -1;
+
+  // Set up AudioLoop object.
+  AudioLoop audio_loop;
+  const size_t kMaxLoopLengthSamples = kSampRateHz * 10;  // 10 second loop.
+  const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000;  // 60 ms.
+  if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+                       kInputBlockSizeSamples))
+    return -1;
+
+  int32_t time_now_ms = 0;
+
+  // Get first input packet.
+  RTPHeader rtp_header;
+  RtpGenerator rtp_gen(kSampRateHz / 1000);
+  // Start with positive drift first half of simulation.
+  rtp_gen.set_drift_factor(drift_factor);
+  bool drift_flipped = false;
+  int32_t packet_input_time_ms =
+      rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
+  auto input_samples = audio_loop.GetNextBlock();
+  if (input_samples.empty())
+    exit(1);
+  uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
+  size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                           input_samples.size(), input_payload);
+  RTC_CHECK_EQ(sizeof(input_payload), payload_len);
+
+  // Main loop.
+  int64_t start_time_ms = clock->TimeInMilliseconds();
+  AudioFrame out_frame;
+  while (time_now_ms < runtime_ms) {
+    while (packet_input_time_ms <= time_now_ms) {
+      // Drop every N packets, where N = FLAG_lossrate.
+      bool lost = false;
+      if (lossrate > 0) {
+        lost = ((rtp_header.sequenceNumber - 1) % lossrate) == 0;
+      }
+      if (!lost) {
+        // Insert packet.
+        int error = neteq->InsertPacket(rtp_header, input_payload);
+        if (error != NetEq::kOK)
+          return -1;
+      }
+
+      // Get next packet.
+      packet_input_time_ms = rtp_gen.GetRtpHeader(
+          kPayloadType, kInputBlockSizeSamples, &rtp_header);
+      input_samples = audio_loop.GetNextBlock();
+      if (input_samples.empty())
+        return -1;
+      payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                        input_samples.size(), input_payload);
+      RTC_DCHECK_EQ(payload_len, kInputBlockSizeSamples * sizeof(int16_t));
+    }
+
+    // Get output audio, but don't do anything with it.
+    bool muted;
+    int error = neteq->GetAudio(&out_frame, &muted);
+    RTC_CHECK(!muted);
+    if (error != NetEq::kOK)
+      return -1;
+
+    RTC_DCHECK_EQ(out_frame.samples_per_channel_, (kSampRateHz * 10) / 1000);
+
+    static const int kOutputBlockSizeMs = 10;
+    time_now_ms += kOutputBlockSizeMs;
+    if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {
+      // Apply negative drift second half of simulation.
+      rtp_gen.set_drift_factor(-drift_factor);
+      drift_flipped = true;
+    }
+  }
+  int64_t end_time_ms = clock->TimeInMilliseconds();
+  return end_time_ms - start_time_ms;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
new file mode 100644
index 0000000000..b5b4d91577
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
@@ -0,0 +1,32 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+namespace test {
+
+class NetEqPerformanceTest {
+ public:
+  // Runs a performance test with parameters as follows:
+  //   `runtime_ms`: the simulation time, i.e., the duration of the audio data.
+  //   `lossrate`: drop one out of `lossrate` packets, e.g., one out of 10.
+  //   `drift_factor`: clock drift in [0, 1].
+  // Returns the runtime in ms.
+  static int64_t Run(int runtime_ms, int lossrate, double drift_factor);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
new file mode 100644
index 0000000000..18e6091f93
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -0,0 +1,482 @@
+/*
+ *  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/neteq/tools/neteq_quality_test.h"
+
+#include <stdio.h>
+
+#include <cmath>
+
+#include "absl/flags/flag.h"
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "modules/audio_coding/neteq/tools/output_audio_file.h"
+#include "modules/audio_coding/neteq/tools/output_wav_file.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/string_encode.h"
+#include "system_wrappers/include/clock.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(std::string,
+          in_filename,
+          "audio_coding/speech_mono_16kHz.pcm",
+          "Path of the input file (relative to the resources/ directory) for "
+          "input audio (specify sample rate with --input_sample_rate, "
+          "and channels with --channels).");
+
+ABSL_FLAG(int, input_sample_rate, 16000, "Sample rate of input file in Hz.");
+
+ABSL_FLAG(int, channels, 1, "Number of channels in input audio.");
+
+ABSL_FLAG(std::string,
+          out_filename,
+          "neteq_quality_test_out.pcm",
+          "Name of output audio file, which will be saved in " +
+              ::webrtc::test::OutputPath());
+
+ABSL_FLAG(
+    int,
+    runtime_ms,
+    10000,
+    "Simulated runtime (milliseconds). -1 will consume the complete file.");
+
+ABSL_FLAG(int, packet_loss_rate, 10, "Percentile of packet loss.");
+
+ABSL_FLAG(int,
+          random_loss_mode,
+          ::webrtc::test::kUniformLoss,
+          "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot "
+          "loss, 3--fixed loss.");
+
+ABSL_FLAG(int,
+          burst_length,
+          30,
+          "Burst length in milliseconds, only valid for Gilbert Elliot loss.");
+
+ABSL_FLAG(float, drift_factor, 0.0, "Time drift factor.");
+
+ABSL_FLAG(int,
+          preload_packets,
+          1,
+          "Preload the buffer with this many packets.");
+
+ABSL_FLAG(std::string,
+          loss_events,
+          "",
+          "List of loss events time and duration separated by comma: "
+          "<first_event_time> <first_event_duration>, <second_event_time> "
+          "<second_event_duration>, ...");
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+std::unique_ptr<NetEq> CreateNetEq(
+    const NetEq::Config& config,
+    Clock* clock,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+  return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock);
+}
+
+const std::string& GetInFilenamePath(absl::string_view file_name) {
+  std::vector<absl::string_view> name_parts = rtc::split(file_name, '.');
+  RTC_CHECK_EQ(name_parts.size(), 2);
+  static const std::string path =
+      ::webrtc::test::ResourcePath(name_parts[0], name_parts[1]);
+  return path;
+}
+
+const std::string& GetOutFilenamePath(absl::string_view file_name) {
+  static const std::string path =
+      ::webrtc::test::OutputPath() + std::string(file_name);
+  return path;
+}
+
+}  // namespace
+
+const uint8_t kPayloadType = 95;
+const int kOutputSizeMs = 10;
+const int kInitSeed = 0x12345678;
+const int kPacketLossTimeUnitMs = 10;
+
+// Common validator for file names.
+static bool ValidateFilename(absl::string_view value, bool is_output) {
+  if (!is_output) {
+    RTC_CHECK_NE(value.substr(value.find_last_of('.') + 1), "wav")
+        << "WAV file input is not supported";
+  }
+  FILE* fid = is_output ? fopen(std::string(value).c_str(), "wb")
+                        : fopen(std::string(value).c_str(), "rb");
+  if (fid == nullptr)
+    return false;
+  fclose(fid);
+  return true;
+}
+
+// ProbTrans00Solver() is to calculate the transition probability from no-loss
+// state to itself in a modified Gilbert Elliot packet loss model. The result is
+// to achieve the target packet loss rate `loss_rate`, when a packet is not
+// lost only if all `units` drawings within the duration of the packet result in
+// no-loss.
+static double ProbTrans00Solver(int units,
+                                double loss_rate,
+                                double prob_trans_10) {
+  if (units == 1)
+    return prob_trans_10 / (1.0f - loss_rate) - prob_trans_10;
+  // 0 == prob_trans_00 ^ (units - 1) + (1 - loss_rate) / prob_trans_10 *
+  //     prob_trans_00 - (1 - loss_rate) * (1 + 1 / prob_trans_10).
+  // There is a unique solution between 0.0 and 1.0, due to the monotonicity and
+  // an opposite sign at 0.0 and 1.0.
+  // For simplicity, we reformulate the equation as
+  //     f(x) = x ^ (units - 1) + a x + b.
+  // Its derivative is
+  //     f'(x) = (units - 1) x ^ (units - 2) + a.
+  // The derivative is strictly greater than 0 when x is between 0 and 1.
+  // We use Newton's method to solve the equation, iteration is
+  //     x(k+1) = x(k) - f(x) / f'(x);
+  const double kPrecision = 0.001f;
+  const int kIterations = 100;
+  const double a = (1.0f - loss_rate) / prob_trans_10;
+  const double b = (loss_rate - 1.0f) * (1.0f + 1.0f / prob_trans_10);
+  double x = 0.0;  // Starting point;
+  double f = b;
+  double f_p;
+  int iter = 0;
+  while ((f >= kPrecision || f <= -kPrecision) && iter < kIterations) {
+    f_p = (units - 1.0f) * std::pow(x, units - 2) + a;
+    x -= f / f_p;
+    if (x > 1.0f) {
+      x = 1.0f;
+    } else if (x < 0.0f) {
+      x = 0.0f;
+    }
+    f = std::pow(x, units - 1) + a * x + b;
+    iter++;
+  }
+  return x;
+}
+
+NetEqQualityTest::NetEqQualityTest(
+    int block_duration_ms,
+    int in_sampling_khz,
+    int out_sampling_khz,
+    const SdpAudioFormat& format,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
+    : audio_format_(format),
+      channels_(absl::GetFlag(FLAGS_channels)),
+      decoded_time_ms_(0),
+      decodable_time_ms_(0),
+      drift_factor_(absl::GetFlag(FLAGS_drift_factor)),
+      packet_loss_rate_(absl::GetFlag(FLAGS_packet_loss_rate)),
+      block_duration_ms_(block_duration_ms),
+      in_sampling_khz_(in_sampling_khz),
+      out_sampling_khz_(out_sampling_khz),
+      in_size_samples_(
+          static_cast<size_t>(in_sampling_khz_ * block_duration_ms_)),
+      payload_size_bytes_(0),
+      max_payload_bytes_(0),
+      in_file_(new ResampleInputAudioFile(
+          GetInFilenamePath(absl::GetFlag(FLAGS_in_filename)),
+          absl::GetFlag(FLAGS_input_sample_rate),
+          in_sampling_khz * 1000,
+          absl::GetFlag(FLAGS_runtime_ms) > 0)),
+      rtp_generator_(
+          new RtpGenerator(in_sampling_khz_, 0, 0, decodable_time_ms_)),
+      total_payload_size_bytes_(0) {
+  // Flag validation
+  RTC_CHECK(ValidateFilename(
+      GetInFilenamePath(absl::GetFlag(FLAGS_in_filename)), false))
+      << "Invalid input filename.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_input_sample_rate) == 8000 ||
+            absl::GetFlag(FLAGS_input_sample_rate) == 16000 ||
+            absl::GetFlag(FLAGS_input_sample_rate) == 32000 ||
+            absl::GetFlag(FLAGS_input_sample_rate) == 48000)
+      << "Invalid sample rate should be 8000, 16000, 32000 or 48000 Hz.";
+
+  RTC_CHECK_EQ(absl::GetFlag(FLAGS_channels), 1)
+      << "Invalid number of channels, current support only 1.";
+
+  RTC_CHECK(ValidateFilename(
+      GetOutFilenamePath(absl::GetFlag(FLAGS_out_filename)), true))
+      << "Invalid output filename.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_packet_loss_rate) >= 0 &&
+            absl::GetFlag(FLAGS_packet_loss_rate) <= 100)
+      << "Invalid packet loss percentile, should be between 0 and 100.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_random_loss_mode) >= 0 &&
+            absl::GetFlag(FLAGS_random_loss_mode) < kLastLossMode)
+      << "Invalid random packet loss mode, should be between 0 and "
+      << kLastLossMode - 1 << ".";
+
+  RTC_CHECK_GE(absl::GetFlag(FLAGS_burst_length), kPacketLossTimeUnitMs)
+      << "Invalid burst length, should be greater than or equal to "
+      << kPacketLossTimeUnitMs << " ms.";
+
+  RTC_CHECK_GT(absl::GetFlag(FLAGS_drift_factor), -0.1)
+      << "Invalid drift factor, should be greater than -0.1.";
+
+  RTC_CHECK_GE(absl::GetFlag(FLAGS_preload_packets), 0)
+      << "Invalid number of packets to preload; must be non-negative.";
+
+  const std::string out_filename =
+      GetOutFilenamePath(absl::GetFlag(FLAGS_out_filename));
+  const std::string log_filename = out_filename + ".log";
+  log_file_.open(log_filename.c_str(), std::ofstream::out);
+  RTC_CHECK(log_file_.is_open());
+
+  if (out_filename.size() >= 4 &&
+      out_filename.substr(out_filename.size() - 4) == ".wav") {
+    // Open a wav file.
+    output_.reset(
+        new webrtc::test::OutputWavFile(out_filename, 1000 * out_sampling_khz));
+  } else {
+    // Open a pcm file.
+    output_.reset(new webrtc::test::OutputAudioFile(out_filename));
+  }
+
+  NetEq::Config config;
+  config.sample_rate_hz = out_sampling_khz_ * 1000;
+  neteq_ = CreateNetEq(config, Clock::GetRealTimeClock(), decoder_factory);
+  max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
+  in_data_.reset(new int16_t[in_size_samples_ * channels_]);
+}
+
+NetEqQualityTest::~NetEqQualityTest() {
+  log_file_.close();
+}
+
+bool NoLoss::Lost(int now_ms) {
+  return false;
+}
+
+UniformLoss::UniformLoss(double loss_rate) : loss_rate_(loss_rate) {}
+
+bool UniformLoss::Lost(int now_ms) {
+  int drop_this = rand();
+  return (drop_this < loss_rate_ * RAND_MAX);
+}
+
+GilbertElliotLoss::GilbertElliotLoss(double prob_trans_11, double prob_trans_01)
+    : prob_trans_11_(prob_trans_11),
+      prob_trans_01_(prob_trans_01),
+      lost_last_(false),
+      uniform_loss_model_(new UniformLoss(0)) {}
+
+GilbertElliotLoss::~GilbertElliotLoss() {}
+
+bool GilbertElliotLoss::Lost(int now_ms) {
+  // Simulate bursty channel (Gilbert model).
+  // (1st order) Markov chain model with memory of the previous/last
+  // packet state (lost or received).
+  if (lost_last_) {
+    // Previous packet was not received.
+    uniform_loss_model_->set_loss_rate(prob_trans_11_);
+    return lost_last_ = uniform_loss_model_->Lost(now_ms);
+  } else {
+    uniform_loss_model_->set_loss_rate(prob_trans_01_);
+    return lost_last_ = uniform_loss_model_->Lost(now_ms);
+  }
+}
+
+FixedLossModel::FixedLossModel(
+    std::set<FixedLossEvent, FixedLossEventCmp> loss_events)
+    : loss_events_(loss_events) {
+  loss_events_it_ = loss_events_.begin();
+}
+
+FixedLossModel::~FixedLossModel() {}
+
+bool FixedLossModel::Lost(int now_ms) {
+  if (loss_events_it_ != loss_events_.end() &&
+      now_ms > loss_events_it_->start_ms) {
+    if (now_ms <= loss_events_it_->start_ms + loss_events_it_->duration_ms) {
+      return true;
+    } else {
+      ++loss_events_it_;
+      return false;
+    }
+  }
+  return false;
+}
+
+void NetEqQualityTest::SetUp() {
+  ASSERT_TRUE(neteq_->RegisterPayloadType(kPayloadType, audio_format_));
+  rtp_generator_->set_drift_factor(drift_factor_);
+
+  int units = block_duration_ms_ / kPacketLossTimeUnitMs;
+  switch (absl::GetFlag(FLAGS_random_loss_mode)) {
+    case kUniformLoss: {
+      // `unit_loss_rate` is the packet loss rate for each unit time interval
+      // (kPacketLossTimeUnitMs). Since a packet loss event is generated if any
+      // of |block_duration_ms_ / kPacketLossTimeUnitMs| unit time intervals of
+      // a full packet duration is drawn with a loss, `unit_loss_rate` fulfills
+      // (1 - unit_loss_rate) ^ (block_duration_ms_ / kPacketLossTimeUnitMs) ==
+      // 1 - packet_loss_rate.
+      double unit_loss_rate =
+          (1.0 - std::pow(1.0 - 0.01 * packet_loss_rate_, 1.0 / units));
+      loss_model_.reset(new UniformLoss(unit_loss_rate));
+      break;
+    }
+    case kGilbertElliotLoss: {
+      // `FLAGS_burst_length` should be integer times of kPacketLossTimeUnitMs.
+      ASSERT_EQ(0, absl::GetFlag(FLAGS_burst_length) % kPacketLossTimeUnitMs);
+
+      // We do not allow 100 percent packet loss in Gilbert Elliot model, which
+      // makes no sense.
+      ASSERT_GT(100, packet_loss_rate_);
+
+      // To guarantee the overall packet loss rate, transition probabilities
+      // need to satisfy:
+      // pi_0 * (1 - prob_trans_01_) ^ units +
+      //     pi_1 * prob_trans_10_ ^ (units - 1) == 1 - loss_rate
+      // pi_0 = prob_trans_10 / (prob_trans_10 + prob_trans_01_)
+      //     is the stationary state probability of no-loss
+      // pi_1 = prob_trans_01_ / (prob_trans_10 + prob_trans_01_)
+      //     is the stationary state probability of loss
+      // After a derivation prob_trans_00 should satisfy:
+      // prob_trans_00 ^ (units - 1) = (loss_rate - 1) / prob_trans_10 *
+      //     prob_trans_00 + (1 - loss_rate) * (1 + 1 / prob_trans_10).
+      double loss_rate = 0.01f * packet_loss_rate_;
+      double prob_trans_10 =
+          1.0f * kPacketLossTimeUnitMs / absl::GetFlag(FLAGS_burst_length);
+      double prob_trans_00 = ProbTrans00Solver(units, loss_rate, prob_trans_10);
+      loss_model_.reset(
+          new GilbertElliotLoss(1.0f - prob_trans_10, 1.0f - prob_trans_00));
+      break;
+    }
+    case kFixedLoss: {
+      std::istringstream loss_events_stream(absl::GetFlag(FLAGS_loss_events));
+      std::string loss_event_string;
+      std::set<FixedLossEvent, FixedLossEventCmp> loss_events;
+      while (std::getline(loss_events_stream, loss_event_string, ',')) {
+        std::vector<int> loss_event_params;
+        std::istringstream loss_event_params_stream(loss_event_string);
+        std::copy(std::istream_iterator<int>(loss_event_params_stream),
+                  std::istream_iterator<int>(),
+                  std::back_inserter(loss_event_params));
+        RTC_CHECK_EQ(loss_event_params.size(), 2);
+        auto result = loss_events.insert(
+            FixedLossEvent(loss_event_params[0], loss_event_params[1]));
+        RTC_CHECK(result.second);
+      }
+      RTC_CHECK_GT(loss_events.size(), 0);
+      loss_model_.reset(new FixedLossModel(loss_events));
+      break;
+    }
+    default: {
+      loss_model_.reset(new NoLoss);
+      break;
+    }
+  }
+
+  // Make sure that the packet loss profile is same for all derived tests.
+  srand(kInitSeed);
+}
+
+std::ofstream& NetEqQualityTest::Log() {
+  return log_file_;
+}
+
+bool NetEqQualityTest::PacketLost() {
+  int cycles = block_duration_ms_ / kPacketLossTimeUnitMs;
+
+  // The loop is to make sure that codecs with different block lengths share the
+  // same packet loss profile.
+  bool lost = false;
+  for (int idx = 0; idx < cycles; idx++) {
+    if (loss_model_->Lost(decoded_time_ms_)) {
+      // The packet will be lost if any of the drawings indicates a loss, but
+      // the loop has to go on to make sure that codecs with different block
+      // lengths keep the same pace.
+      lost = true;
+    }
+  }
+  return lost;
+}
+
+int NetEqQualityTest::Transmit() {
+  int packet_input_time_ms = rtp_generator_->GetRtpHeader(
+      kPayloadType, in_size_samples_, &rtp_header_);
+  Log() << "Packet of size " << payload_size_bytes_ << " bytes, for frame at "
+        << packet_input_time_ms << " ms ";
+  if (payload_size_bytes_ > 0) {
+    if (!PacketLost()) {
+      int ret = neteq_->InsertPacket(
+          rtp_header_,
+          rtc::ArrayView<const uint8_t>(payload_.data(), payload_size_bytes_));
+      if (ret != NetEq::kOK)
+        return -1;
+      Log() << "was sent.";
+    } else {
+      Log() << "was lost.";
+    }
+  }
+  Log() << std::endl;
+  return packet_input_time_ms;
+}
+
+int NetEqQualityTest::DecodeBlock() {
+  bool muted;
+  int ret = neteq_->GetAudio(&out_frame_, &muted);
+  RTC_CHECK(!muted);
+
+  if (ret != NetEq::kOK) {
+    return -1;
+  } else {
+    RTC_DCHECK_EQ(out_frame_.num_channels_, channels_);
+    RTC_DCHECK_EQ(out_frame_.samples_per_channel_,
+                  static_cast<size_t>(kOutputSizeMs * out_sampling_khz_));
+    RTC_CHECK(output_->WriteArray(
+        out_frame_.data(),
+        out_frame_.samples_per_channel_ * out_frame_.num_channels_));
+    return static_cast<int>(out_frame_.samples_per_channel_);
+  }
+}
+
+void NetEqQualityTest::Simulate() {
+  int audio_size_samples;
+  bool end_of_input = false;
+  int runtime_ms = absl::GetFlag(FLAGS_runtime_ms) >= 0
+                       ? absl::GetFlag(FLAGS_runtime_ms)
+                       : INT_MAX;
+
+  while (!end_of_input && decoded_time_ms_ < runtime_ms) {
+    // Preload the buffer if needed.
+    while (decodable_time_ms_ -
+               absl::GetFlag(FLAGS_preload_packets) * block_duration_ms_ <
+           decoded_time_ms_) {
+      if (!in_file_->Read(in_size_samples_ * channels_, &in_data_[0])) {
+        end_of_input = true;
+        ASSERT_TRUE(end_of_input && absl::GetFlag(FLAGS_runtime_ms) < 0);
+        break;
+      }
+      payload_.Clear();
+      payload_size_bytes_ = EncodeBlock(&in_data_[0], in_size_samples_,
+                                        &payload_, max_payload_bytes_);
+      total_payload_size_bytes_ += payload_size_bytes_;
+      decodable_time_ms_ = Transmit() + block_duration_ms_;
+    }
+    audio_size_samples = DecodeBlock();
+    if (audio_size_samples > 0) {
+      decoded_time_ms_ += audio_size_samples / out_sampling_khz_;
+    }
+  }
+  Log() << "Average bit rate was "
+        << 8.0f * total_payload_size_bytes_ / absl::GetFlag(FLAGS_runtime_ms)
+        << " kbps" << std::endl;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
new file mode 100644
index 0000000000..edcb117748
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -0,0 +1,176 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
+
+#include <fstream>
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+enum LossModes {
+  kNoLoss,
+  kUniformLoss,
+  kGilbertElliotLoss,
+  kFixedLoss,
+  kLastLossMode
+};
+
+class LossModel {
+ public:
+  virtual ~LossModel() {}
+  virtual bool Lost(int now_ms) = 0;
+};
+
+class NoLoss : public LossModel {
+ public:
+  bool Lost(int now_ms) override;
+};
+
+class UniformLoss : public LossModel {
+ public:
+  UniformLoss(double loss_rate);
+  bool Lost(int now_ms) override;
+  void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; }
+
+ private:
+  double loss_rate_;
+};
+
+class GilbertElliotLoss : public LossModel {
+ public:
+  GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
+  ~GilbertElliotLoss() override;
+  bool Lost(int now_ms) override;
+
+ private:
+  // Prob. of losing current packet, when previous packet is lost.
+  double prob_trans_11_;
+  // Prob. of losing current packet, when previous packet is not lost.
+  double prob_trans_01_;
+  bool lost_last_;
+  std::unique_ptr<UniformLoss> uniform_loss_model_;
+};
+
+struct FixedLossEvent {
+  int start_ms;
+  int duration_ms;
+  FixedLossEvent(int start_ms, int duration_ms)
+      : start_ms(start_ms), duration_ms(duration_ms) {}
+};
+
+struct FixedLossEventCmp {
+  bool operator()(const FixedLossEvent& l_event,
+                  const FixedLossEvent& r_event) const {
+    return l_event.start_ms < r_event.start_ms;
+  }
+};
+
+class FixedLossModel : public LossModel {
+ public:
+  FixedLossModel(std::set<FixedLossEvent, FixedLossEventCmp> loss_events);
+  ~FixedLossModel() override;
+  bool Lost(int now_ms) override;
+
+ private:
+  std::set<FixedLossEvent, FixedLossEventCmp> loss_events_;
+  std::set<FixedLossEvent, FixedLossEventCmp>::iterator loss_events_it_;
+};
+
+class NetEqQualityTest : public ::testing::Test {
+ protected:
+  NetEqQualityTest(
+      int block_duration_ms,
+      int in_sampling_khz,
+      int out_sampling_khz,
+      const SdpAudioFormat& format,
+      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory =
+          webrtc::CreateBuiltinAudioDecoderFactory());
+  ~NetEqQualityTest() override;
+
+  void SetUp() override;
+
+  // EncodeBlock(...) does the following:
+  // 1. encodes a block of audio, saved in `in_data` and has a length of
+  // `block_size_samples` (samples per channel),
+  // 2. save the bit stream to `payload` of `max_bytes` bytes in size,
+  // 3. returns the length of the payload (in bytes),
+  virtual int EncodeBlock(int16_t* in_data,
+                          size_t block_size_samples,
+                          rtc::Buffer* payload,
+                          size_t max_bytes) = 0;
+
+  // PacketLost(...) determines weather a packet sent at an indicated time gets
+  // lost or not.
+  bool PacketLost();
+
+  // DecodeBlock() decodes a block of audio using the payload stored in
+  // `payload_` with the length of `payload_size_bytes_` (bytes). The decoded
+  // audio is to be stored in `out_data_`.
+  int DecodeBlock();
+
+  // Transmit() uses `rtp_generator_` to generate a packet and passes it to
+  // `neteq_`.
+  int Transmit();
+
+  // Runs encoding / transmitting / decoding.
+  void Simulate();
+
+  // Write to log file. Usage Log() << ...
+  std::ofstream& Log();
+
+  SdpAudioFormat audio_format_;
+  const size_t channels_;
+
+ private:
+  int decoded_time_ms_;
+  int decodable_time_ms_;
+  double drift_factor_;
+  int packet_loss_rate_;
+  const int block_duration_ms_;
+  const int in_sampling_khz_;
+  const int out_sampling_khz_;
+
+  // Number of samples per channel in a frame.
+  const size_t in_size_samples_;
+
+  size_t payload_size_bytes_;
+  size_t max_payload_bytes_;
+
+  std::unique_ptr<InputAudioFile> in_file_;
+  std::unique_ptr<AudioSink> output_;
+  std::ofstream log_file_;
+
+  std::unique_ptr<RtpGenerator> rtp_generator_;
+  std::unique_ptr<NetEq> neteq_;
+  std::unique_ptr<LossModel> loss_model_;
+
+  std::unique_ptr<int16_t[]> in_data_;
+  rtc::Buffer payload_;
+  AudioFrame out_frame_;
+  RTPHeader rtp_header_;
+
+  size_t total_payload_size_bytes_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
new file mode 100644
index 0000000000..ffd114ae5b
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
@@ -0,0 +1,116 @@
+/*
+ *  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/neteq/tools/neteq_replacement_input.h"
+
+#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqReplacementInput::NetEqReplacementInput(
+    std::unique_ptr<NetEqInput> source,
+    uint8_t replacement_payload_type,
+    const std::set<uint8_t>& comfort_noise_types,
+    const std::set<uint8_t>& forbidden_types)
+    : source_(std::move(source)),
+      replacement_payload_type_(replacement_payload_type),
+      comfort_noise_types_(comfort_noise_types),
+      forbidden_types_(forbidden_types) {
+  RTC_CHECK(source_);
+  packet_ = source_->PopPacket();
+  ReplacePacket();
+}
+
+absl::optional<int64_t> NetEqReplacementInput::NextPacketTime() const {
+  return packet_
+             ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms))
+             : absl::nullopt;
+}
+
+absl::optional<int64_t> NetEqReplacementInput::NextOutputEventTime() const {
+  return source_->NextOutputEventTime();
+}
+
+std::unique_ptr<NetEqInput::PacketData> NetEqReplacementInput::PopPacket() {
+  std::unique_ptr<PacketData> to_return = std::move(packet_);
+  while (true) {
+    packet_ = source_->PopPacket();
+    if (!packet_)
+      break;
+    if (packet_->payload.size() > packet_->header.paddingLength) {
+      // Not padding only. Good to go. Skip this packet otherwise.
+      break;
+    }
+  }
+  ReplacePacket();
+  return to_return;
+}
+
+void NetEqReplacementInput::AdvanceOutputEvent() {
+  source_->AdvanceOutputEvent();
+}
+
+bool NetEqReplacementInput::ended() const {
+  return source_->ended();
+}
+
+absl::optional<RTPHeader> NetEqReplacementInput::NextHeader() const {
+  return source_->NextHeader();
+}
+
+void NetEqReplacementInput::ReplacePacket() {
+  if (!source_->NextPacketTime()) {
+    // End of input. Cannot do proper replacement on the very last packet, so we
+    // delete it instead.
+    packet_.reset();
+    return;
+  }
+
+  RTC_DCHECK(packet_);
+
+  RTC_CHECK_EQ(forbidden_types_.count(packet_->header.payloadType), 0)
+      << "Payload type " << static_cast<int>(packet_->header.payloadType)
+      << " is forbidden.";
+
+  // Check if this packet is comfort noise.
+  if (comfort_noise_types_.count(packet_->header.payloadType) != 0) {
+    // If CNG, simply insert a zero-energy one-byte payload.
+    uint8_t cng_payload[1] = {127};  // Max attenuation of CNG.
+    packet_->payload.SetData(cng_payload);
+    return;
+  }
+
+  absl::optional<RTPHeader> next_hdr = source_->NextHeader();
+  RTC_DCHECK(next_hdr);
+  uint8_t payload[12];
+  RTC_DCHECK_LE(last_frame_size_timestamps_, 120 * 48);
+  uint32_t input_frame_size_timestamps = last_frame_size_timestamps_;
+  const uint32_t timestamp_diff =
+      next_hdr->timestamp - packet_->header.timestamp;
+  if (next_hdr->sequenceNumber == packet_->header.sequenceNumber + 1 &&
+      timestamp_diff <= 120 * 48) {
+    // Packets are in order and the timestamp diff is less than 5760 samples.
+    // Accept the timestamp diff as a valid frame size.
+    input_frame_size_timestamps = timestamp_diff;
+    last_frame_size_timestamps_ = input_frame_size_timestamps;
+  }
+  RTC_DCHECK_LE(input_frame_size_timestamps, 120 * 48);
+  FakeDecodeFromFile::PrepareEncoded(packet_->header.timestamp,
+                                     input_frame_size_timestamps,
+                                     packet_->payload.size(), payload);
+  packet_->payload.SetData(payload);
+  packet_->header.payloadType = replacement_payload_type_;
+  return;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h
new file mode 100644
index 0000000000..9ce9b9dc63
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h
@@ -0,0 +1,51 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
+
+#include <memory>
+#include <set>
+
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+
+namespace webrtc {
+namespace test {
+
+// This class converts the packets from a NetEqInput to fake encodings to be
+// decoded by a FakeDecodeFromFile decoder.
+class NetEqReplacementInput : public NetEqInput {
+ public:
+  NetEqReplacementInput(std::unique_ptr<NetEqInput> source,
+                        uint8_t replacement_payload_type,
+                        const std::set<uint8_t>& comfort_noise_types,
+                        const std::set<uint8_t>& forbidden_types);
+
+  absl::optional<int64_t> NextPacketTime() const override;
+  absl::optional<int64_t> NextOutputEventTime() const override;
+  std::unique_ptr<PacketData> PopPacket() override;
+  void AdvanceOutputEvent() override;
+  bool ended() const override;
+  absl::optional<RTPHeader> NextHeader() const override;
+
+ private:
+  void ReplacePacket();
+
+  std::unique_ptr<NetEqInput> source_;
+  const uint8_t replacement_payload_type_;
+  const std::set<uint8_t> comfort_noise_types_;
+  const std::set<uint8_t> forbidden_types_;
+  std::unique_ptr<PacketData> packet_;         // The next packet to deliver.
+  uint32_t last_frame_size_timestamps_ = 960;  // Initial guess: 20 ms @ 48 kHz.
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
new file mode 100644
index 0000000000..b274069bd4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -0,0 +1,406 @@
+/*
+ *  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 <iostream>
+#include <string>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "modules/audio_coding/neteq/tools/neteq_test_factory.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/field_trial.h"
+
+using TestConfig = webrtc::test::NetEqTestFactory::Config;
+
+ABSL_FLAG(bool,
+          codec_map,
+          false,
+          "Prints the mapping between RTP payload type and "
+          "codec");
+ABSL_FLAG(std::string,
+          force_fieldtrials,
+          "",
+          "Field trials control experimental feature code which can be forced. "
+          "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
+          " will assign the group Enable to field trial WebRTC-FooFeature.");
+ABSL_FLAG(int, pcmu, TestConfig::default_pcmu(), "RTP payload type for PCM-u");
+ABSL_FLAG(int, pcma, TestConfig::default_pcma(), "RTP payload type for PCM-a");
+ABSL_FLAG(int, ilbc, TestConfig::default_ilbc(), "RTP payload type for iLBC");
+ABSL_FLAG(int, isac, TestConfig::default_isac(), "RTP payload type for iSAC");
+ABSL_FLAG(int,
+          isac_swb,
+          TestConfig::default_isac_swb(),
+          "RTP payload type for iSAC-swb (32 kHz)");
+ABSL_FLAG(int, opus, TestConfig::default_opus(), "RTP payload type for Opus");
+ABSL_FLAG(int,
+          pcm16b,
+          TestConfig::default_pcm16b(),
+          "RTP payload type for PCM16b-nb (8 kHz)");
+ABSL_FLAG(int,
+          pcm16b_wb,
+          TestConfig::default_pcm16b_wb(),
+          "RTP payload type for PCM16b-wb (16 kHz)");
+ABSL_FLAG(int,
+          pcm16b_swb32,
+          TestConfig::default_pcm16b_swb32(),
+          "RTP payload type for PCM16b-swb32 (32 kHz)");
+ABSL_FLAG(int,
+          pcm16b_swb48,
+          TestConfig::default_pcm16b_swb48(),
+          "RTP payload type for PCM16b-swb48 (48 kHz)");
+ABSL_FLAG(int, g722, TestConfig::default_g722(), "RTP payload type for G.722");
+ABSL_FLAG(int,
+          avt,
+          TestConfig::default_avt(),
+          "RTP payload type for AVT/DTMF (8 kHz)");
+ABSL_FLAG(int,
+          avt_16,
+          TestConfig::default_avt_16(),
+          "RTP payload type for AVT/DTMF (16 kHz)");
+ABSL_FLAG(int,
+          avt_32,
+          TestConfig::default_avt_32(),
+          "RTP payload type for AVT/DTMF (32 kHz)");
+ABSL_FLAG(int,
+          avt_48,
+          TestConfig::default_avt_48(),
+          "RTP payload type for AVT/DTMF (48 kHz)");
+ABSL_FLAG(int,
+          red,
+          TestConfig::default_red(),
+          "RTP payload type for redundant audio (RED)");
+ABSL_FLAG(int,
+          cn_nb,
+          TestConfig::default_cn_nb(),
+          "RTP payload type for comfort noise (8 kHz)");
+ABSL_FLAG(int,
+          cn_wb,
+          TestConfig::default_cn_wb(),
+          "RTP payload type for comfort noise (16 kHz)");
+ABSL_FLAG(int,
+          cn_swb32,
+          TestConfig::default_cn_swb32(),
+          "RTP payload type for comfort noise (32 kHz)");
+ABSL_FLAG(int,
+          cn_swb48,
+          TestConfig::default_cn_swb48(),
+          "RTP payload type for comfort noise (48 kHz)");
+ABSL_FLAG(std::string,
+          replacement_audio_file,
+          "",
+          "A PCM file that will be used to populate dummy"
+          " RTP packets");
+ABSL_FLAG(std::string,
+          ssrc,
+          "",
+          "Only use packets with this SSRC (decimal or hex, the latter "
+          "starting with 0x)");
+ABSL_FLAG(int,
+          audio_level,
+          TestConfig::default_audio_level(),
+          "Extension ID for audio level (RFC 6464)");
+ABSL_FLAG(int,
+          abs_send_time,
+          TestConfig::default_abs_send_time(),
+          "Extension ID for absolute sender time");
+ABSL_FLAG(int,
+          transport_seq_no,
+          TestConfig::default_transport_seq_no(),
+          "Extension ID for transport sequence number");
+ABSL_FLAG(int,
+          video_content_type,
+          TestConfig::default_video_content_type(),
+          "Extension ID for video content type");
+ABSL_FLAG(int,
+          video_timing,
+          TestConfig::default_video_timing(),
+          "Extension ID for video timing");
+ABSL_FLAG(std::string,
+          output_files_base_name,
+          "",
+          "Custom path used as prefix for the output files - i.e., "
+          "matlab plot, python plot, text log.");
+ABSL_FLAG(bool,
+          matlabplot,
+          false,
+          "Generates a matlab script for plotting the delay profile");
+ABSL_FLAG(bool,
+          pythonplot,
+          false,
+          "Generates a python script for plotting the delay profile");
+ABSL_FLAG(bool,
+          textlog,
+          false,
+          "Generates a text log describing the simulation on a "
+          "step-by-step basis.");
+ABSL_FLAG(bool, concealment_events, false, "Prints concealment events");
+ABSL_FLAG(int,
+          max_nr_packets_in_buffer,
+          TestConfig::default_max_nr_packets_in_buffer(),
+          "Maximum allowed number of packets in the buffer");
+ABSL_FLAG(bool,
+          enable_fast_accelerate,
+          false,
+          "Enables jitter buffer fast accelerate");
+
+namespace {
+
+// Parses the input string for a valid SSRC (at the start of the string). If a
+// valid SSRC is found, it is written to the output variable `ssrc`, and true is
+// returned. Otherwise, false is returned.
+bool ParseSsrc(absl::string_view str, uint32_t* ssrc) {
+  if (str.empty())
+    return true;
+  int base = 10;
+  // Look for "0x" or "0X" at the start and change base to 16 if found.
+  if ((str.compare(0, 2, "0x") == 0) || (str.compare(0, 2, "0X") == 0))
+    base = 16;
+  errno = 0;
+  char* end_ptr;
+  std::string str_str = std::string(str);
+  unsigned long value = strtoul(str_str.c_str(), &end_ptr, base);  // NOLINT
+  if (value == ULONG_MAX && errno == ERANGE)
+    return false;  // Value out of range for unsigned long.
+  if (sizeof(unsigned long) > sizeof(uint32_t) && value > 0xFFFFFFFF)  // NOLINT
+    return false;  // Value out of range for uint32_t.
+  if (end_ptr - str_str.c_str() < static_cast<ptrdiff_t>(str.length()))
+    return false;  // Part of the string was not parsed.
+  *ssrc = static_cast<uint32_t>(value);
+  return true;
+}
+
+static bool ValidateExtensionId(int value) {
+  if (value > 0 && value <= 255)  // Value is ok.
+    return true;
+  printf("Extension ID must be between 1 and 255, not %d\n",
+         static_cast<int>(value));
+  return false;
+}
+
+// Flag validators.
+bool ValidatePayloadType(int value) {
+  if (value >= 0 && value <= 127)  // Value is ok.
+    return true;
+  printf("Payload type must be between 0 and 127, not %d\n",
+         static_cast<int>(value));
+  return false;
+}
+
+bool ValidateSsrcValue(absl::string_view str) {
+  uint32_t dummy_ssrc;
+  if (ParseSsrc(str, &dummy_ssrc))  // Value is ok.
+    return true;
+  printf("Invalid SSRC: %.*s\n", static_cast<int>(str.size()), str.data());
+  return false;
+}
+
+void PrintCodecMappingEntry(absl::string_view codec, int flag) {
+  std::cout << codec << ": " << flag << std::endl;
+}
+
+void PrintCodecMapping() {
+  PrintCodecMappingEntry("PCM-u", absl::GetFlag(FLAGS_pcmu));
+  PrintCodecMappingEntry("PCM-a", absl::GetFlag(FLAGS_pcma));
+  PrintCodecMappingEntry("iLBC", absl::GetFlag(FLAGS_ilbc));
+  PrintCodecMappingEntry("iSAC", absl::GetFlag(FLAGS_isac));
+  PrintCodecMappingEntry("iSAC-swb (32 kHz)", absl::GetFlag(FLAGS_isac_swb));
+  PrintCodecMappingEntry("Opus", absl::GetFlag(FLAGS_opus));
+  PrintCodecMappingEntry("PCM16b-nb (8 kHz)", absl::GetFlag(FLAGS_pcm16b));
+  PrintCodecMappingEntry("PCM16b-wb (16 kHz)", absl::GetFlag(FLAGS_pcm16b_wb));
+  PrintCodecMappingEntry("PCM16b-swb32 (32 kHz)",
+                         absl::GetFlag(FLAGS_pcm16b_swb32));
+  PrintCodecMappingEntry("PCM16b-swb48 (48 kHz)",
+                         absl::GetFlag(FLAGS_pcm16b_swb48));
+  PrintCodecMappingEntry("G.722", absl::GetFlag(FLAGS_g722));
+  PrintCodecMappingEntry("AVT/DTMF (8 kHz)", absl::GetFlag(FLAGS_avt));
+  PrintCodecMappingEntry("AVT/DTMF (16 kHz)", absl::GetFlag(FLAGS_avt_16));
+  PrintCodecMappingEntry("AVT/DTMF (32 kHz)", absl::GetFlag(FLAGS_avt_32));
+  PrintCodecMappingEntry("AVT/DTMF (48 kHz)", absl::GetFlag(FLAGS_avt_48));
+  PrintCodecMappingEntry("redundant audio (RED)", absl::GetFlag(FLAGS_red));
+  PrintCodecMappingEntry("comfort noise (8 kHz)", absl::GetFlag(FLAGS_cn_nb));
+  PrintCodecMappingEntry("comfort noise (16 kHz)", absl::GetFlag(FLAGS_cn_wb));
+  PrintCodecMappingEntry("comfort noise (32 kHz)",
+                         absl::GetFlag(FLAGS_cn_swb32));
+  PrintCodecMappingEntry("comfort noise (48 kHz)",
+                         absl::GetFlag(FLAGS_cn_swb48));
+}
+
+bool ValidateOutputFilesOptions(bool textlog,
+                                bool plotting,
+                                absl::string_view output_files_base_name,
+                                absl::string_view output_audio_filename) {
+  bool output_files_base_name_specified = !output_files_base_name.empty();
+  if (!textlog && !plotting && output_files_base_name_specified) {
+    std::cout << "Error: --output_files_base_name cannot be used without at "
+                 "least one of the following flags: --textlog, --matlabplot, "
+                 "--pythonplot."
+              << std::endl;
+    return false;
+  }
+  // Without `output_audio_filename`, `output_files_base_name` is required when
+  // plotting output files must be generated (in order to form a valid output
+  // file name).
+  if (output_audio_filename.empty() && plotting &&
+      !output_files_base_name_specified) {
+    std::cout << "Error: when no output audio file is specified and "
+                 "--matlabplot and/or --pythonplot are used, "
+                 "--output_files_base_name must be also used."
+              << std::endl;
+    return false;
+  }
+  return true;
+}
+
+absl::optional<std::string> CreateOptionalOutputFileName(
+    bool output_requested,
+    absl::string_view basename,
+    absl::string_view output_audio_filename,
+    absl::string_view suffix) {
+  if (!output_requested) {
+    return absl::nullopt;
+  }
+  if (!basename.empty()) {
+    // Override the automatic assignment.
+    rtc::StringBuilder sb(basename);
+    sb << suffix;
+    return sb.str();
+  }
+  if (!output_audio_filename.empty()) {
+    // Automatically assign name.
+    rtc::StringBuilder sb(output_audio_filename);
+    sb << suffix;
+    return sb.str();
+  }
+  std::cout << "Error: invalid text log file parameters.";
+  return absl::nullopt;
+}
+
+}  // namespace
+
+int main(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  webrtc::test::NetEqTestFactory factory;
+  std::string usage =
+      "Tool for decoding an RTP dump file using NetEq.\n"
+      "Example usage:\n"
+      "./neteq_rtpplay input.rtp [output.{pcm, wav}]\n";
+  if (absl::GetFlag(FLAGS_codec_map)) {
+    PrintCodecMapping();
+    exit(0);
+  }
+  if (args.size() != 2 &&
+      args.size() != 3) {  // The output audio file is optional.
+    // Print usage information.
+    std::cout << usage;
+    exit(0);
+  }
+  const std::string output_audio_filename((args.size() == 3) ? args[2] : "");
+  const std::string output_files_base_name(
+      absl::GetFlag(FLAGS_output_files_base_name));
+  RTC_CHECK(ValidateOutputFilesOptions(
+      absl::GetFlag(FLAGS_textlog),
+      absl::GetFlag(FLAGS_matlabplot) || absl::GetFlag(FLAGS_pythonplot),
+      output_files_base_name, output_audio_filename));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcmu)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcma)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_ilbc)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_isac)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_isac_swb)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_opus)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_wb)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_swb32)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_pcm16b_swb48)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_g722)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt_16)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt_32)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_avt_48)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_red)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_nb)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_wb)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_swb32)));
+  RTC_CHECK(ValidatePayloadType(absl::GetFlag(FLAGS_cn_swb48)));
+  RTC_CHECK(ValidateSsrcValue(absl::GetFlag(FLAGS_ssrc)));
+  RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_audio_level)));
+  RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_abs_send_time)));
+  RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_transport_seq_no)));
+  RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_video_content_type)));
+  RTC_CHECK(ValidateExtensionId(absl::GetFlag(FLAGS_video_timing)));
+
+  // Make force_fieldtrials persistent string during entire program live as
+  // absl::GetFlag creates temporary string and c_str() will point to
+  // deallocated string.
+  const std::string force_fieldtrials = absl::GetFlag(FLAGS_force_fieldtrials);
+  webrtc::field_trial::InitFieldTrialsFromString(force_fieldtrials.c_str());
+
+  webrtc::test::NetEqTestFactory::Config config;
+  config.pcmu = absl::GetFlag(FLAGS_pcmu);
+  config.pcma = absl::GetFlag(FLAGS_pcma);
+  config.ilbc = absl::GetFlag(FLAGS_ilbc);
+  config.isac = absl::GetFlag(FLAGS_isac);
+  config.isac_swb = absl::GetFlag(FLAGS_isac_swb);
+  config.opus = absl::GetFlag(FLAGS_opus);
+  config.pcm16b = absl::GetFlag(FLAGS_pcm16b);
+  config.pcm16b_wb = absl::GetFlag(FLAGS_pcm16b_wb);
+  config.pcm16b_swb32 = absl::GetFlag(FLAGS_pcm16b_swb32);
+  config.pcm16b_swb48 = absl::GetFlag(FLAGS_pcm16b_swb48);
+  config.g722 = absl::GetFlag(FLAGS_g722);
+  config.avt = absl::GetFlag(FLAGS_avt);
+  config.avt_16 = absl::GetFlag(FLAGS_avt_16);
+  config.avt_32 = absl::GetFlag(FLAGS_avt_32);
+  config.avt_48 = absl::GetFlag(FLAGS_avt_48);
+  config.red = absl::GetFlag(FLAGS_red);
+  config.cn_nb = absl::GetFlag(FLAGS_cn_nb);
+  config.cn_wb = absl::GetFlag(FLAGS_cn_wb);
+  config.cn_swb32 = absl::GetFlag(FLAGS_cn_swb32);
+  config.cn_swb48 = absl::GetFlag(FLAGS_cn_swb48);
+  config.replacement_audio_file = absl::GetFlag(FLAGS_replacement_audio_file);
+  config.audio_level = absl::GetFlag(FLAGS_audio_level);
+  config.abs_send_time = absl::GetFlag(FLAGS_abs_send_time);
+  config.transport_seq_no = absl::GetFlag(FLAGS_transport_seq_no);
+  config.video_content_type = absl::GetFlag(FLAGS_video_content_type);
+  config.video_timing = absl::GetFlag(FLAGS_video_timing);
+  config.matlabplot = absl::GetFlag(FLAGS_matlabplot);
+  config.pythonplot = absl::GetFlag(FLAGS_pythonplot);
+  config.concealment_events = absl::GetFlag(FLAGS_concealment_events);
+  config.max_nr_packets_in_buffer =
+      absl::GetFlag(FLAGS_max_nr_packets_in_buffer);
+  config.enable_fast_accelerate = absl::GetFlag(FLAGS_enable_fast_accelerate);
+  if (!output_audio_filename.empty()) {
+    config.output_audio_filename = output_audio_filename;
+  }
+  config.textlog = absl::GetFlag(FLAGS_textlog);
+  config.textlog_filename = CreateOptionalOutputFileName(
+      absl::GetFlag(FLAGS_textlog), output_files_base_name,
+      output_audio_filename, ".text_log.txt");
+  config.plot_scripts_basename = CreateOptionalOutputFileName(
+      absl::GetFlag(FLAGS_matlabplot) || absl::GetFlag(FLAGS_pythonplot),
+      output_files_base_name, output_audio_filename, "");
+
+  // Check if an SSRC value was provided.
+  if (absl::GetFlag(FLAGS_ssrc).size() > 0) {
+    uint32_t ssrc;
+    RTC_CHECK(ParseSsrc(absl::GetFlag(FLAGS_ssrc), &ssrc))
+        << "Flag verification has failed.";
+    config.ssrc_filter = absl::make_optional(ssrc);
+  }
+
+  std::unique_ptr<webrtc::test::NetEqTest> test =
+      factory.InitializeTestFromFile(/*input_filename=*/args[1],
+                                     /*factory=*/nullptr, config);
+  RTC_CHECK(test) << "ERROR: Unable to run test";
+  test->Run();
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh
new file mode 100755
index 0000000000..0a6bf16016
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_rtpplay_test.sh
@@ -0,0 +1,183 @@
+#!/bin/bash
+#
+#  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.
+#
+
+# Aliases.
+BIN=$1
+TEST_RTC_EVENT_LOG=$2
+INPUT_PCM_FILE=$3
+
+# Check setup.
+if [ ! -f $BIN ]; then
+  echo "Cannot find neteq_rtpplay binary."
+  exit 99
+fi
+if [ ! -f $TEST_RTC_EVENT_LOG ]; then
+  echo "Cannot find RTC event log file."
+  exit 99
+fi
+if [ ! -f $INPUT_PCM_FILE ]; then
+  echo "Cannot find PCM file."
+  exit 99
+fi
+
+# Defines.
+
+TMP_DIR=$(mktemp -d /tmp/tmp_XXXXXXXXXX)
+PASS=0
+FAIL=1
+TEST_SUITE_RESULT=$PASS
+
+file_hash () {
+  md5sum $1 | awk '{ print $1 }'
+}
+
+test_passed () {
+  echo PASS
+}
+
+test_failed () {
+  echo "FAIL: $1"
+  TEST_SUITE_RESULT=$FAIL
+}
+
+test_file_checksums_match () {
+  if [ ! -f $1 ] || [ ! -f $2 ]; then
+    test_failed "Cannot compare hash values: file(s) not found."
+    return
+  fi
+  HASH1=$(file_hash $1)
+  HASH2=$(file_hash $2)
+  if [ "$HASH1" = "$HASH2" ]; then
+    test_passed
+  else
+    test_failed "$1 differs from $2"
+  fi
+}
+
+test_file_exists () {
+  if [ -f $1 ]; then
+    test_passed
+  else
+    test_failed "$1 does not exist"
+  fi
+}
+
+test_exit_code_0 () {
+  if [ $1 -eq 0 ]; then
+    test_passed
+  else
+    test_failed "$1 did not return 0"
+  fi
+}
+
+test_exit_code_not_0 () {
+  if [ $1 -eq 0 ]; then
+    test_failed "$1 returned 0"
+  else
+    test_passed
+  fi
+}
+
+# Generate test data.
+
+# Case 1. Pre-existing way.
+CASE1_WAV=$TMP_DIR/case1.wav
+$BIN $TEST_RTC_EVENT_LOG $CASE1_WAV  \
+    --replacement_audio_file $INPUT_PCM_FILE  \
+    --textlog --pythonplot --matlabplot  \
+    > $TMP_DIR/case1.stdout 2> /dev/null
+CASE1_RETURN_CODE=$?
+CASE1_TEXTLOG=$TMP_DIR/case1.wav.text_log.txt
+CASE1_PYPLOT=$TMP_DIR/case1_wav.py
+CASE1_MATPLOT=$TMP_DIR/case1_wav.m
+
+# Case 2. No output files.
+$BIN $TEST_RTC_EVENT_LOG --replacement_audio_file $INPUT_PCM_FILE  \
+    > $TMP_DIR/case2.stdout 2> /dev/null
+CASE2_RETURN_CODE=$?
+
+# Case 3. No output audio file.
+
+# Case 3.1 Without --output_files_base_name (won't run).
+$BIN $TEST_RTC_EVENT_LOG  \
+    --replacement_audio_file $INPUT_PCM_FILE  \
+    --textlog --pythonplot --matlabplot  \
+    &> /dev/null
+CASE3_1_RETURN_CODE=$?
+
+# Case 3.2 With --output_files_base_name (runs).
+$BIN $TEST_RTC_EVENT_LOG  \
+    --replacement_audio_file $INPUT_PCM_FILE  \
+    --output_files_base_name $TMP_DIR/case3_2  \
+    --textlog --pythonplot --matlabplot  \
+    > $TMP_DIR/case3_2.stdout 2> /dev/null
+CASE3_2_RETURN_CODE=$?
+CASE3_2_TEXTLOG=$TMP_DIR/case3_2.text_log.txt
+CASE3_2_PYPLOT=$TMP_DIR/case3_2.py
+CASE3_2_MATPLOT=$TMP_DIR/case3_2.m
+
+# Case 4. With output audio file and --output_files_base_name.
+CASE4_WAV=$TMP_DIR/case4.wav
+$BIN $TEST_RTC_EVENT_LOG $TMP_DIR/case4.wav \
+    --replacement_audio_file $INPUT_PCM_FILE  \
+    --output_files_base_name $TMP_DIR/case4  \
+    --textlog --pythonplot --matlabplot  \
+    > $TMP_DIR/case4.stdout 2> /dev/null
+CASE4_RETURN_CODE=$?
+CASE4_TEXTLOG=$TMP_DIR/case4.text_log.txt
+CASE4_PYPLOT=$TMP_DIR/case4.py
+CASE4_MATPLOT=$TMP_DIR/case4.m
+
+# Tests.
+
+echo Check exit codes
+test_exit_code_0 $CASE1_RETURN_CODE
+test_exit_code_0 $CASE2_RETURN_CODE
+test_exit_code_not_0 $CASE3_1_RETURN_CODE
+test_exit_code_0 $CASE3_2_RETURN_CODE
+test_exit_code_0 $CASE4_RETURN_CODE
+
+echo Check that the expected output files exist
+test_file_exists $CASE1_TEXTLOG
+test_file_exists $CASE3_2_TEXTLOG
+test_file_exists $CASE4_TEXTLOG
+test_file_exists $CASE1_PYPLOT
+test_file_exists $CASE3_2_PYPLOT
+test_file_exists $CASE4_PYPLOT
+test_file_exists $CASE1_MATPLOT
+test_file_exists $CASE3_2_MATPLOT
+test_file_exists $CASE4_MATPLOT
+
+echo Check that the same WAV file is produced
+test_file_checksums_match $CASE1_WAV $CASE4_WAV
+
+echo Check that the same text log is produced
+test_file_checksums_match $CASE1_TEXTLOG $CASE3_2_TEXTLOG
+test_file_checksums_match $CASE1_TEXTLOG $CASE4_TEXTLOG
+
+echo Check that the same python plot scripts is produced
+test_file_checksums_match $CASE1_PYPLOT $CASE3_2_PYPLOT
+test_file_checksums_match $CASE1_PYPLOT $CASE4_PYPLOT
+
+echo Check that the same matlab plot scripts is produced
+test_file_checksums_match $CASE1_MATPLOT $CASE3_2_MATPLOT
+test_file_checksums_match $CASE1_MATPLOT $CASE4_MATPLOT
+
+# Clean up
+rm -fr $TMP_DIR
+
+if [ $TEST_SUITE_RESULT -eq $PASS ]; then
+  echo All tests passed.
+  exit 0
+else
+  echo One or more tests failed.
+  exit 1
+fi
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.cc
new file mode 100644
index 0000000000..6738e494f6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.cc
@@ -0,0 +1,139 @@
+/*
+ *  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/neteq/tools/neteq_stats_getter.h"
+
+#include <algorithm>
+#include <numeric>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+namespace test {
+
+std::string NetEqStatsGetter::ConcealmentEvent::ToString() const {
+  char ss_buf[256];
+  rtc::SimpleStringBuilder ss(ss_buf);
+  ss << "ConcealmentEvent duration_ms:" << duration_ms
+     << " event_number:" << concealment_event_number
+     << " time_from_previous_event_end_ms:" << time_from_previous_event_end_ms;
+  return ss.str();
+}
+
+NetEqStatsGetter::NetEqStatsGetter(
+    std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer)
+    : delay_analyzer_(std::move(delay_analyzer)) {}
+
+void NetEqStatsGetter::BeforeGetAudio(NetEq* neteq) {
+  if (delay_analyzer_) {
+    delay_analyzer_->BeforeGetAudio(neteq);
+  }
+}
+
+void NetEqStatsGetter::AfterGetAudio(int64_t time_now_ms,
+                                     const AudioFrame& audio_frame,
+                                     bool muted,
+                                     NetEq* neteq) {
+  // TODO(minyue): Get stats should better not be called as a call back after
+  // get audio. It is called independently from get audio in practice.
+  const auto lifetime_stat = neteq->GetLifetimeStatistics();
+  if (last_stats_query_time_ms_ == 0 ||
+      rtc::TimeDiff(time_now_ms, last_stats_query_time_ms_) >=
+          stats_query_interval_ms_) {
+    NetEqNetworkStatistics stats;
+    RTC_CHECK_EQ(neteq->NetworkStatistics(&stats), 0);
+    stats_.push_back(std::make_pair(time_now_ms, stats));
+    lifetime_stats_.push_back(std::make_pair(time_now_ms, lifetime_stat));
+    last_stats_query_time_ms_ = time_now_ms;
+  }
+
+  const auto voice_concealed_samples =
+      lifetime_stat.concealed_samples - lifetime_stat.silent_concealed_samples;
+  if (current_concealment_event_ != lifetime_stat.concealment_events &&
+      voice_concealed_samples_until_last_event_ < voice_concealed_samples) {
+    if (last_event_end_time_ms_ > 0) {
+      // Do not account for the first event to avoid start of the call
+      // skewing.
+      ConcealmentEvent concealment_event;
+      uint64_t last_event_voice_concealed_samples =
+          voice_concealed_samples - voice_concealed_samples_until_last_event_;
+      RTC_CHECK_GT(last_event_voice_concealed_samples, 0);
+      concealment_event.duration_ms = last_event_voice_concealed_samples /
+                                      (audio_frame.sample_rate_hz_ / 1000);
+      concealment_event.concealment_event_number = current_concealment_event_;
+      concealment_event.time_from_previous_event_end_ms =
+          time_now_ms - last_event_end_time_ms_;
+      concealment_events_.emplace_back(concealment_event);
+      voice_concealed_samples_until_last_event_ = voice_concealed_samples;
+    }
+    last_event_end_time_ms_ = time_now_ms;
+    voice_concealed_samples_until_last_event_ = voice_concealed_samples;
+    current_concealment_event_ = lifetime_stat.concealment_events;
+  }
+
+  if (delay_analyzer_) {
+    delay_analyzer_->AfterGetAudio(time_now_ms, audio_frame, muted, neteq);
+  }
+}
+
+double NetEqStatsGetter::AverageSpeechExpandRate() const {
+  double sum_speech_expand = std::accumulate(
+      stats_.begin(), stats_.end(), double{0.0},
+      [](double a, std::pair<int64_t, NetEqNetworkStatistics> b) {
+        return a + static_cast<double>(b.second.speech_expand_rate);
+      });
+  return sum_speech_expand / 16384.0 / stats_.size();
+}
+
+NetEqStatsGetter::Stats NetEqStatsGetter::AverageStats() const {
+  Stats sum_stats = std::accumulate(
+      stats_.begin(), stats_.end(), Stats(),
+      [](Stats a, std::pair<int64_t, NetEqNetworkStatistics> bb) {
+        const auto& b = bb.second;
+        a.current_buffer_size_ms += b.current_buffer_size_ms;
+        a.preferred_buffer_size_ms += b.preferred_buffer_size_ms;
+        a.jitter_peaks_found += b.jitter_peaks_found;
+        a.expand_rate += b.expand_rate / 16384.0;
+        a.speech_expand_rate += b.speech_expand_rate / 16384.0;
+        a.preemptive_rate += b.preemptive_rate / 16384.0;
+        a.accelerate_rate += b.accelerate_rate / 16384.0;
+        a.secondary_decoded_rate += b.secondary_decoded_rate / 16384.0;
+        a.secondary_discarded_rate += b.secondary_discarded_rate / 16384.0;
+        a.mean_waiting_time_ms += b.mean_waiting_time_ms;
+        a.median_waiting_time_ms += b.median_waiting_time_ms;
+        a.min_waiting_time_ms = std::min(
+            a.min_waiting_time_ms, static_cast<double>(b.min_waiting_time_ms));
+        a.max_waiting_time_ms = std::max(
+            a.max_waiting_time_ms, static_cast<double>(b.max_waiting_time_ms));
+        return a;
+      });
+
+  sum_stats.current_buffer_size_ms /= stats_.size();
+  sum_stats.preferred_buffer_size_ms /= stats_.size();
+  sum_stats.jitter_peaks_found /= stats_.size();
+  sum_stats.packet_loss_rate /= stats_.size();
+  sum_stats.expand_rate /= stats_.size();
+  sum_stats.speech_expand_rate /= stats_.size();
+  sum_stats.preemptive_rate /= stats_.size();
+  sum_stats.accelerate_rate /= stats_.size();
+  sum_stats.secondary_decoded_rate /= stats_.size();
+  sum_stats.secondary_discarded_rate /= stats_.size();
+  sum_stats.added_zero_samples /= stats_.size();
+  sum_stats.mean_waiting_time_ms /= stats_.size();
+  sum_stats.median_waiting_time_ms /= stats_.size();
+
+  return sum_stats;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h
new file mode 100644
index 0000000000..b1b12bb1f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h
@@ -0,0 +1,106 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqStatsGetter : public NetEqGetAudioCallback {
+ public:
+  // This struct is a replica of webrtc::NetEqNetworkStatistics, but with all
+  // values stored in double precision.
+  struct Stats {
+    double current_buffer_size_ms = 0.0;
+    double preferred_buffer_size_ms = 0.0;
+    double jitter_peaks_found = 0.0;
+    double packet_loss_rate = 0.0;
+    double expand_rate = 0.0;
+    double speech_expand_rate = 0.0;
+    double preemptive_rate = 0.0;
+    double accelerate_rate = 0.0;
+    double secondary_decoded_rate = 0.0;
+    double secondary_discarded_rate = 0.0;
+    double clockdrift_ppm = 0.0;
+    double added_zero_samples = 0.0;
+    double mean_waiting_time_ms = 0.0;
+    double median_waiting_time_ms = 0.0;
+    double min_waiting_time_ms = 0.0;
+    double max_waiting_time_ms = 0.0;
+  };
+
+  struct ConcealmentEvent {
+    uint64_t duration_ms;
+    size_t concealment_event_number;
+    int64_t time_from_previous_event_end_ms;
+    std::string ToString() const;
+  };
+
+  // Takes a pointer to another callback object, which will be invoked after
+  // this object finishes. This does not transfer ownership, and null is a
+  // valid value.
+  explicit NetEqStatsGetter(std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer);
+
+  void set_stats_query_interval_ms(int64_t stats_query_interval_ms) {
+    stats_query_interval_ms_ = stats_query_interval_ms;
+  }
+
+  void BeforeGetAudio(NetEq* neteq) override;
+
+  void AfterGetAudio(int64_t time_now_ms,
+                     const AudioFrame& audio_frame,
+                     bool muted,
+                     NetEq* neteq) override;
+
+  double AverageSpeechExpandRate() const;
+
+  NetEqDelayAnalyzer* delay_analyzer() const { return delay_analyzer_.get(); }
+
+  const std::vector<ConcealmentEvent>& concealment_events() const {
+    // Do not account for the last concealment event to avoid potential end
+    // call skewing.
+    return concealment_events_;
+  }
+
+  const std::vector<std::pair<int64_t, NetEqNetworkStatistics>>* stats() const {
+    return &stats_;
+  }
+
+  const std::vector<std::pair<int64_t, NetEqLifetimeStatistics>>*
+  lifetime_stats() const {
+    return &lifetime_stats_;
+  }
+
+  Stats AverageStats() const;
+
+ private:
+  std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer_;
+  int64_t stats_query_interval_ms_ = 1000;
+  int64_t last_stats_query_time_ms_ = 0;
+  std::vector<std::pair<int64_t, NetEqNetworkStatistics>> stats_;
+  std::vector<std::pair<int64_t, NetEqLifetimeStatistics>> lifetime_stats_;
+  size_t current_concealment_event_ = 1;
+  uint64_t voice_concealed_samples_until_last_event_ = 0;
+  std::vector<ConcealmentEvent> concealment_events_;
+  int64_t last_event_end_time_ms_ = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
new file mode 100644
index 0000000000..162a4c9300
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.cc
@@ -0,0 +1,111 @@
+/*
+ *  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/neteq/tools/neteq_stats_plotter.h"
+
+#include <inttypes.h>
+#include <stdio.h>
+
+#include <utility>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqStatsPlotter::NetEqStatsPlotter(bool make_matlab_plot,
+                                     bool make_python_plot,
+                                     bool show_concealment_events,
+                                     absl::string_view base_file_name)
+    : make_matlab_plot_(make_matlab_plot),
+      make_python_plot_(make_python_plot),
+      show_concealment_events_(show_concealment_events),
+      base_file_name_(base_file_name) {
+  std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer;
+  if (make_matlab_plot || make_python_plot) {
+    delay_analyzer.reset(new NetEqDelayAnalyzer);
+  }
+  stats_getter_.reset(new NetEqStatsGetter(std::move(delay_analyzer)));
+}
+
+void NetEqStatsPlotter::SimulationEnded(int64_t simulation_time_ms) {
+  if (make_matlab_plot_) {
+    auto matlab_script_name = base_file_name_;
+    std::replace(matlab_script_name.begin(), matlab_script_name.end(), '.',
+                 '_');
+    printf("Creating Matlab plot script %s.m\n", matlab_script_name.c_str());
+    stats_getter_->delay_analyzer()->CreateMatlabScript(matlab_script_name +
+                                                        ".m");
+  }
+  if (make_python_plot_) {
+    auto python_script_name = base_file_name_;
+    std::replace(python_script_name.begin(), python_script_name.end(), '.',
+                 '_');
+    printf("Creating Python plot script %s.py\n", python_script_name.c_str());
+    stats_getter_->delay_analyzer()->CreatePythonScript(python_script_name +
+                                                        ".py");
+  }
+
+  printf("Simulation statistics:\n");
+  printf("  output duration: %" PRId64 " ms\n", simulation_time_ms);
+  auto stats = stats_getter_->AverageStats();
+  printf("  packet_loss_rate: %f %%\n", 100.0 * stats.packet_loss_rate);
+  printf("  expand_rate: %f %%\n", 100.0 * stats.expand_rate);
+  printf("  speech_expand_rate: %f %%\n", 100.0 * stats.speech_expand_rate);
+  printf("  preemptive_rate: %f %%\n", 100.0 * stats.preemptive_rate);
+  printf("  accelerate_rate: %f %%\n", 100.0 * stats.accelerate_rate);
+  printf("  secondary_decoded_rate: %f %%\n",
+         100.0 * stats.secondary_decoded_rate);
+  printf("  secondary_discarded_rate: %f %%\n",
+         100.0 * stats.secondary_discarded_rate);
+  printf("  clockdrift_ppm: %f ppm\n", stats.clockdrift_ppm);
+  printf("  mean_waiting_time_ms: %f ms\n", stats.mean_waiting_time_ms);
+  printf("  median_waiting_time_ms: %f ms\n", stats.median_waiting_time_ms);
+  printf("  min_waiting_time_ms: %f ms\n", stats.min_waiting_time_ms);
+  printf("  max_waiting_time_ms: %f ms\n", stats.max_waiting_time_ms);
+  printf("  current_buffer_size_ms: %f ms\n", stats.current_buffer_size_ms);
+  printf("  preferred_buffer_size_ms: %f ms\n", stats.preferred_buffer_size_ms);
+  if (show_concealment_events_) {
+    printf(" concealment_events_ms:\n");
+    for (auto concealment_event : stats_getter_->concealment_events())
+      printf("%s\n", concealment_event.ToString().c_str());
+    printf(" end of concealment_events_ms\n");
+  }
+
+  const auto lifetime_stats_vector = stats_getter_->lifetime_stats();
+  if (!lifetime_stats_vector->empty()) {
+    auto lifetime_stats = lifetime_stats_vector->back().second;
+    printf("  total_samples_received: %" PRIu64 "\n",
+           lifetime_stats.total_samples_received);
+    printf("  concealed_samples: %" PRIu64 "\n",
+           lifetime_stats.concealed_samples);
+    printf("  concealment_events: %" PRIu64 "\n",
+           lifetime_stats.concealment_events);
+    printf("  delayed_packet_outage_samples: %" PRIu64 "\n",
+           lifetime_stats.delayed_packet_outage_samples);
+    printf("  num_interruptions: %d\n", lifetime_stats.interruption_count);
+    printf("  sum_interruption_length_ms: %d ms\n",
+           lifetime_stats.total_interruption_duration_ms);
+    printf("  interruption_ratio: %f\n",
+           static_cast<double>(lifetime_stats.total_interruption_duration_ms) /
+               simulation_time_ms);
+    printf("  removed_samples_for_acceleration: %" PRIu64 "\n",
+           lifetime_stats.removed_samples_for_acceleration);
+    printf("  inserted_samples_for_deceleration: %" PRIu64 "\n",
+           lifetime_stats.inserted_samples_for_deceleration);
+    printf("  generated_noise_samples: %" PRIu64 "\n",
+           lifetime_stats.generated_noise_samples);
+    printf("  packets_discarded: %" PRIu64 "\n",
+           lifetime_stats.packets_discarded);
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.h
new file mode 100644
index 0000000000..11c16da9d1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_stats_plotter.h
@@ -0,0 +1,47 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqStatsPlotter : public NetEqSimulationEndedCallback {
+ public:
+  NetEqStatsPlotter(bool make_matlab_plot,
+                    bool make_python_plot,
+                    bool show_concealment_events,
+                    absl::string_view base_file_name);
+
+  void SimulationEnded(int64_t simulation_time_ms) override;
+
+  NetEqStatsGetter* stats_getter() { return stats_getter_.get(); }
+
+ private:
+  std::unique_ptr<NetEqStatsGetter> stats_getter_;
+  const bool make_matlab_plot_;
+  const bool make_python_plot_;
+  const bool show_concealment_events_;
+  const std::string base_file_name_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_PLOTTER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.cc
new file mode 100644
index 0000000000..a567efe2de
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -0,0 +1,345 @@
+/*
+ *  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/neteq/tools/neteq_test.h"
+
+#include <iomanip>
+#include <iostream>
+
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+absl::optional<NetEq::Operation> ActionToOperations(
+    absl::optional<NetEqSimulator::Action> a) {
+  if (!a) {
+    return absl::nullopt;
+  }
+  switch (*a) {
+    case NetEqSimulator::Action::kAccelerate:
+      return absl::make_optional(NetEq::Operation::kAccelerate);
+    case NetEqSimulator::Action::kExpand:
+      return absl::make_optional(NetEq::Operation::kExpand);
+    case NetEqSimulator::Action::kNormal:
+      return absl::make_optional(NetEq::Operation::kNormal);
+    case NetEqSimulator::Action::kPreemptiveExpand:
+      return absl::make_optional(NetEq::Operation::kPreemptiveExpand);
+  }
+}
+
+std::unique_ptr<NetEq> CreateNetEq(
+    const NetEq::Config& config,
+    Clock* clock,
+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+  return DefaultNetEqFactory().CreateNetEq(config, decoder_factory, clock);
+}
+
+}  // namespace
+
+void DefaultNetEqTestErrorCallback::OnInsertPacketError(
+    const NetEqInput::PacketData& packet) {
+  std::cerr << "InsertPacket returned an error." << std::endl;
+  std::cerr << "Packet data: " << packet.ToString() << std::endl;
+  RTC_FATAL();
+}
+
+void DefaultNetEqTestErrorCallback::OnGetAudioError() {
+  std::cerr << "GetAudio returned an error." << std::endl;
+  RTC_FATAL();
+}
+
+NetEqTest::NetEqTest(const NetEq::Config& config,
+                     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+                     const DecoderMap& codecs,
+                     std::unique_ptr<std::ofstream> text_log,
+                     NetEqFactory* neteq_factory,
+                     std::unique_ptr<NetEqInput> input,
+                     std::unique_ptr<AudioSink> output,
+                     Callbacks callbacks)
+    : input_(std::move(input)),
+      clock_(Timestamp::Millis(input_->NextEventTime().value_or(0))),
+      neteq_(neteq_factory
+                 ? neteq_factory->CreateNetEq(config, decoder_factory, &clock_)
+                 : CreateNetEq(config, &clock_, decoder_factory)),
+      output_(std::move(output)),
+      callbacks_(callbacks),
+      sample_rate_hz_(config.sample_rate_hz),
+      text_log_(std::move(text_log)) {
+  RTC_CHECK(!config.enable_muted_state)
+      << "The code does not handle enable_muted_state";
+  RegisterDecoders(codecs);
+}
+
+NetEqTest::~NetEqTest() = default;
+
+int64_t NetEqTest::Run() {
+  int64_t simulation_time = 0;
+  SimulationStepResult step_result;
+  do {
+    step_result = RunToNextGetAudio();
+    simulation_time += step_result.simulation_step_ms;
+  } while (!step_result.is_simulation_finished);
+  if (callbacks_.simulation_ended_callback) {
+    callbacks_.simulation_ended_callback->SimulationEnded(simulation_time);
+  }
+  return simulation_time;
+}
+
+NetEqTest::SimulationStepResult NetEqTest::RunToNextGetAudio() {
+  SimulationStepResult result;
+  const int64_t start_time_ms = *input_->NextEventTime();
+  int64_t time_now_ms = clock_.CurrentTime().ms();
+  current_state_.packet_iat_ms.clear();
+
+  while (!input_->ended()) {
+    // Advance time to next event.
+    RTC_DCHECK(input_->NextEventTime());
+    clock_.AdvanceTimeMilliseconds(*input_->NextEventTime() - time_now_ms);
+    time_now_ms = *input_->NextEventTime();
+    // Check if it is time to insert packet.
+    if (input_->NextPacketTime() && time_now_ms >= *input_->NextPacketTime()) {
+      std::unique_ptr<NetEqInput::PacketData> packet_data = input_->PopPacket();
+      RTC_CHECK(packet_data);
+      const size_t payload_data_length =
+          packet_data->payload.size() - packet_data->header.paddingLength;
+      if (payload_data_length != 0) {
+        int error = neteq_->InsertPacket(
+            packet_data->header,
+            rtc::ArrayView<const uint8_t>(packet_data->payload));
+        if (error != NetEq::kOK && callbacks_.error_callback) {
+          callbacks_.error_callback->OnInsertPacketError(*packet_data);
+        }
+        if (callbacks_.post_insert_packet) {
+          callbacks_.post_insert_packet->AfterInsertPacket(*packet_data,
+                                                           neteq_.get());
+        }
+      } else {
+        neteq_->InsertEmptyPacket(packet_data->header);
+      }
+      if (last_packet_time_ms_) {
+        current_state_.packet_iat_ms.push_back(time_now_ms -
+                                               *last_packet_time_ms_);
+      }
+      if (text_log_) {
+        const auto ops_state = neteq_->GetOperationsAndState();
+        const auto delta_wallclock =
+            last_packet_time_ms_ ? (time_now_ms - *last_packet_time_ms_) : -1;
+        const auto delta_timestamp =
+            last_packet_timestamp_
+                ? (static_cast<int64_t>(packet_data->header.timestamp) -
+                   *last_packet_timestamp_) *
+                      1000 / sample_rate_hz_
+                : -1;
+        const auto packet_size_bytes =
+            packet_data->payload.size() == 12
+                ? ByteReader<uint32_t>::ReadLittleEndian(
+                      &packet_data->payload[8])
+                : -1;
+        *text_log_ << "Packet   - wallclock: " << std::setw(5) << time_now_ms
+                   << ", delta wc: " << std::setw(4) << delta_wallclock
+                   << ", seq_no: " << packet_data->header.sequenceNumber
+                   << ", timestamp: " << std::setw(10)
+                   << packet_data->header.timestamp
+                   << ", delta ts: " << std::setw(4) << delta_timestamp
+                   << ", size: " << std::setw(5) << packet_size_bytes
+                   << ", frame size: " << std::setw(3)
+                   << ops_state.current_frame_size_ms
+                   << ", buffer size: " << std::setw(4)
+                   << ops_state.current_buffer_size_ms << std::endl;
+      }
+      last_packet_time_ms_ = absl::make_optional<int>(time_now_ms);
+      last_packet_timestamp_ =
+          absl::make_optional<uint32_t>(packet_data->header.timestamp);
+    }
+
+    // Check if it is time to get output audio.
+    if (input_->NextOutputEventTime() &&
+        time_now_ms >= *input_->NextOutputEventTime()) {
+      if (callbacks_.get_audio_callback) {
+        callbacks_.get_audio_callback->BeforeGetAudio(neteq_.get());
+      }
+      AudioFrame out_frame;
+      bool muted;
+      int error = neteq_->GetAudio(&out_frame, &muted, nullptr,
+                                   ActionToOperations(next_action_));
+      next_action_ = absl::nullopt;
+      RTC_CHECK(!muted) << "The code does not handle enable_muted_state";
+      if (error != NetEq::kOK) {
+        if (callbacks_.error_callback) {
+          callbacks_.error_callback->OnGetAudioError();
+        }
+      } else {
+        sample_rate_hz_ = out_frame.sample_rate_hz_;
+      }
+      if (callbacks_.get_audio_callback) {
+        callbacks_.get_audio_callback->AfterGetAudio(time_now_ms, out_frame,
+                                                     muted, neteq_.get());
+      }
+
+      if (output_) {
+        RTC_CHECK(output_->WriteArray(
+            out_frame.data(),
+            out_frame.samples_per_channel_ * out_frame.num_channels_));
+      }
+
+      input_->AdvanceOutputEvent();
+      result.simulation_step_ms =
+          input_->NextEventTime().value_or(time_now_ms) - start_time_ms;
+      const auto operations_state = neteq_->GetOperationsAndState();
+      current_state_.current_delay_ms = operations_state.current_buffer_size_ms;
+      current_state_.packet_size_ms = operations_state.current_frame_size_ms;
+      current_state_.next_packet_available =
+          operations_state.next_packet_available;
+      current_state_.packet_buffer_flushed =
+          operations_state.packet_buffer_flushes >
+          prev_ops_state_.packet_buffer_flushes;
+      // TODO(ivoc): Add more accurate reporting by tracking the origin of
+      // samples in the sync buffer.
+      result.action_times_ms[Action::kExpand] = 0;
+      result.action_times_ms[Action::kAccelerate] = 0;
+      result.action_times_ms[Action::kPreemptiveExpand] = 0;
+      result.action_times_ms[Action::kNormal] = 0;
+
+      if (out_frame.speech_type_ == AudioFrame::SpeechType::kPLC ||
+          out_frame.speech_type_ == AudioFrame::SpeechType::kPLCCNG) {
+        // Consider the whole frame to be the result of expansion.
+        result.action_times_ms[Action::kExpand] = 10;
+      } else if (operations_state.accelerate_samples -
+                     prev_ops_state_.accelerate_samples >
+                 0) {
+        // Consider the whole frame to be the result of acceleration.
+        result.action_times_ms[Action::kAccelerate] = 10;
+      } else if (operations_state.preemptive_samples -
+                     prev_ops_state_.preemptive_samples >
+                 0) {
+        // Consider the whole frame to be the result of preemptive expansion.
+        result.action_times_ms[Action::kPreemptiveExpand] = 10;
+      } else {
+        // Consider the whole frame to be the result of normal playout.
+        result.action_times_ms[Action::kNormal] = 10;
+      }
+      auto lifetime_stats = LifetimeStats();
+      if (text_log_) {
+        const bool plc =
+            (out_frame.speech_type_ == AudioFrame::SpeechType::kPLC) ||
+            (out_frame.speech_type_ == AudioFrame::SpeechType::kPLCCNG);
+        const bool cng = out_frame.speech_type_ == AudioFrame::SpeechType::kCNG;
+        const bool voice_concealed =
+            (lifetime_stats.concealed_samples -
+             lifetime_stats.silent_concealed_samples) >
+            (prev_lifetime_stats_.concealed_samples -
+             prev_lifetime_stats_.silent_concealed_samples);
+        *text_log_ << "GetAudio - wallclock: " << std::setw(5) << time_now_ms
+                   << ", delta wc: " << std::setw(4)
+                   << (input_->NextEventTime().value_or(time_now_ms) -
+                       start_time_ms)
+                   << ", CNG: " << cng << ", PLC: " << plc
+                   << ", voice concealed: " << voice_concealed
+                   << ", buffer size: " << std::setw(4)
+                   << current_state_.current_delay_ms << std::endl;
+        if (lifetime_stats.packets_discarded >
+            prev_lifetime_stats_.packets_discarded) {
+          *text_log_ << "Discarded "
+                     << (lifetime_stats.packets_discarded -
+                         prev_lifetime_stats_.packets_discarded)
+                     << " primary packets." << std::endl;
+        }
+        if (operations_state.packet_buffer_flushes >
+            prev_ops_state_.packet_buffer_flushes) {
+          *text_log_ << "Flushed packet buffer "
+                     << (operations_state.packet_buffer_flushes -
+                         prev_ops_state_.packet_buffer_flushes)
+                     << " times." << std::endl;
+        }
+      }
+      prev_lifetime_stats_ = lifetime_stats;
+      const bool no_more_packets_to_decode =
+          !input_->NextPacketTime() && !operations_state.next_packet_available;
+      // End the simulation if the gap is too large. This indicates an issue
+      // with the event log file.
+      const bool simulation_step_too_large = result.simulation_step_ms > 1000;
+      if (simulation_step_too_large) {
+        // If we don't reset the step time, the large gap will be included in
+        // the simulation time, which can be a large distortion.
+        result.simulation_step_ms = 10;
+      }
+      result.is_simulation_finished = simulation_step_too_large ||
+                                      no_more_packets_to_decode ||
+                                      input_->ended();
+      prev_ops_state_ = operations_state;
+      return result;
+    }
+  }
+  result.simulation_step_ms =
+      input_->NextEventTime().value_or(time_now_ms) - start_time_ms;
+  result.is_simulation_finished = true;
+  return result;
+}
+
+void NetEqTest::SetNextAction(NetEqTest::Action next_operation) {
+  next_action_ = absl::optional<Action>(next_operation);
+}
+
+NetEqTest::NetEqState NetEqTest::GetNetEqState() {
+  return current_state_;
+}
+
+NetEqNetworkStatistics NetEqTest::SimulationStats() {
+  NetEqNetworkStatistics stats;
+  RTC_CHECK_EQ(neteq_->NetworkStatistics(&stats), 0);
+  return stats;
+}
+
+NetEqLifetimeStatistics NetEqTest::LifetimeStats() const {
+  return neteq_->GetLifetimeStatistics();
+}
+
+NetEqTest::DecoderMap NetEqTest::StandardDecoderMap() {
+  DecoderMap codecs = {
+    {0, SdpAudioFormat("pcmu", 8000, 1)},
+    {8, SdpAudioFormat("pcma", 8000, 1)},
+#ifdef WEBRTC_CODEC_ILBC
+    {102, SdpAudioFormat("ilbc", 8000, 1)},
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+    {111, SdpAudioFormat("opus", 48000, 2)},
+#endif
+    {93, SdpAudioFormat("l16", 8000, 1)},
+    {94, SdpAudioFormat("l16", 16000, 1)},
+    {95, SdpAudioFormat("l16", 32000, 1)},
+    {96, SdpAudioFormat("l16", 48000, 1)},
+    {9, SdpAudioFormat("g722", 8000, 1)},
+    {106, SdpAudioFormat("telephone-event", 8000, 1)},
+    {114, SdpAudioFormat("telephone-event", 16000, 1)},
+    {115, SdpAudioFormat("telephone-event", 32000, 1)},
+    {116, SdpAudioFormat("telephone-event", 48000, 1)},
+    {117, SdpAudioFormat("red", 8000, 1)},
+    {13, SdpAudioFormat("cn", 8000, 1)},
+    {98, SdpAudioFormat("cn", 16000, 1)},
+    {99, SdpAudioFormat("cn", 32000, 1)},
+    {100, SdpAudioFormat("cn", 48000, 1)}
+  };
+  return codecs;
+}
+
+void NetEqTest::RegisterDecoders(const DecoderMap& codecs) {
+  for (const auto& c : codecs) {
+    RTC_CHECK(neteq_->RegisterPayloadType(c.first, c.second))
+        << "Cannot register " << c.second.name << " to payload type "
+        << c.first;
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.h
new file mode 100644
index 0000000000..1d3eeda453
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test.h
@@ -0,0 +1,129 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
+
+#include <fstream>
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/test/neteq_simulator.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace test {
+
+class NetEqTestErrorCallback {
+ public:
+  virtual ~NetEqTestErrorCallback() = default;
+  virtual void OnInsertPacketError(const NetEqInput::PacketData& packet) {}
+  virtual void OnGetAudioError() {}
+};
+
+class DefaultNetEqTestErrorCallback : public NetEqTestErrorCallback {
+  void OnInsertPacketError(const NetEqInput::PacketData& packet) override;
+  void OnGetAudioError() override;
+};
+
+class NetEqPostInsertPacket {
+ public:
+  virtual ~NetEqPostInsertPacket() = default;
+  virtual void AfterInsertPacket(const NetEqInput::PacketData& packet,
+                                 NetEq* neteq) = 0;
+};
+
+class NetEqGetAudioCallback {
+ public:
+  virtual ~NetEqGetAudioCallback() = default;
+  virtual void BeforeGetAudio(NetEq* neteq) = 0;
+  virtual void AfterGetAudio(int64_t time_now_ms,
+                             const AudioFrame& audio_frame,
+                             bool muted,
+                             NetEq* neteq) = 0;
+};
+
+class NetEqSimulationEndedCallback {
+ public:
+  virtual ~NetEqSimulationEndedCallback() = default;
+  virtual void SimulationEnded(int64_t simulation_time_ms) = 0;
+};
+
+// Class that provides an input--output test for NetEq. The input (both packets
+// and output events) is provided by a NetEqInput object, while the output is
+// directed to an AudioSink object.
+class NetEqTest : public NetEqSimulator {
+ public:
+  using DecoderMap = std::map<int, SdpAudioFormat>;
+
+  struct Callbacks {
+    NetEqTestErrorCallback* error_callback = nullptr;
+    NetEqPostInsertPacket* post_insert_packet = nullptr;
+    NetEqGetAudioCallback* get_audio_callback = nullptr;
+    NetEqSimulationEndedCallback* simulation_ended_callback = nullptr;
+  };
+
+  // Sets up the test with given configuration, codec mappings, input, ouput,
+  // and callback objects for error reporting.
+  NetEqTest(const NetEq::Config& config,
+            rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+            const DecoderMap& codecs,
+            std::unique_ptr<std::ofstream> text_log,
+            NetEqFactory* neteq_factory,
+            std::unique_ptr<NetEqInput> input,
+            std::unique_ptr<AudioSink> output,
+            Callbacks callbacks);
+
+  ~NetEqTest() override;
+
+  // Runs the test. Returns the duration of the produced audio in ms.
+  int64_t Run() override;
+  // Runs the simulation until we hit the next GetAudio event. If the simulation
+  // is finished, is_simulation_finished will be set to true in the returned
+  // SimulationStepResult.
+  SimulationStepResult RunToNextGetAudio() override;
+
+  void SetNextAction(Action next_operation) override;
+  NetEqState GetNetEqState() override;
+
+  // Returns the statistics from NetEq.
+  NetEqNetworkStatistics SimulationStats();
+  NetEqLifetimeStatistics LifetimeStats() const;
+
+  static DecoderMap StandardDecoderMap();
+
+ private:
+  void RegisterDecoders(const DecoderMap& codecs);
+  std::unique_ptr<NetEqInput> input_;
+  SimulatedClock clock_;
+  absl::optional<Action> next_action_;
+  absl::optional<int> last_packet_time_ms_;
+  std::unique_ptr<NetEq> neteq_;
+  std::unique_ptr<AudioSink> output_;
+  Callbacks callbacks_;
+  int sample_rate_hz_;
+  NetEqState current_state_;
+  NetEqOperationsAndState prev_ops_state_;
+  NetEqLifetimeStatistics prev_lifetime_stats_;
+  absl::optional<uint32_t> last_packet_timestamp_;
+  std::unique_ptr<std::ofstream> text_log_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc
new file mode 100644
index 0000000000..6cd371406c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.cc
@@ -0,0 +1,342 @@
+/*
+ *  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/neteq/tools/neteq_test_factory.h"
+
+#include <errno.h>
+#include <limits.h>  // For ULONG_MAX returned by strtoul.
+#include <stdio.h>
+#include <stdlib.h>  // For strtoul.
+
+#include <fstream>
+#include <iostream>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+#include "modules/audio_coding/neteq/tools/initial_packet_inserter_neteq_input.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+#include "modules/audio_coding/neteq/tools/neteq_event_log_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
+#include "modules/audio_coding/neteq/tools/neteq_stats_plotter.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "modules/audio_coding/neteq/tools/output_audio_file.h"
+#include "modules/audio_coding/neteq/tools/output_wav_file.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "rtc_base/checks.h"
+#include "test/function_audio_decoder_factory.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+absl::optional<int> CodecSampleRate(
+    uint8_t payload_type,
+    webrtc::test::NetEqTestFactory::Config config) {
+  if (payload_type == config.pcmu || payload_type == config.pcma ||
+      payload_type == config.ilbc || payload_type == config.pcm16b ||
+      payload_type == config.cn_nb || payload_type == config.avt)
+    return 8000;
+  if (payload_type == config.isac || payload_type == config.pcm16b_wb ||
+      payload_type == config.g722 || payload_type == config.cn_wb ||
+      payload_type == config.avt_16)
+    return 16000;
+  if (payload_type == config.isac_swb || payload_type == config.pcm16b_swb32 ||
+      payload_type == config.cn_swb32 || payload_type == config.avt_32)
+    return 32000;
+  if (payload_type == config.opus || payload_type == config.pcm16b_swb48 ||
+      payload_type == config.cn_swb48 || payload_type == config.avt_48)
+    return 48000;
+  if (payload_type == config.red)
+    return 0;
+  return absl::nullopt;
+}
+
+}  // namespace
+
+// A callback class which prints whenver the inserted packet stream changes
+// the SSRC.
+class SsrcSwitchDetector : public NetEqPostInsertPacket {
+ public:
+  // Takes a pointer to another callback object, which will be invoked after
+  // this object finishes. This does not transfer ownership, and null is a
+  // valid value.
+  explicit SsrcSwitchDetector(NetEqPostInsertPacket* other_callback)
+      : other_callback_(other_callback) {}
+
+  void AfterInsertPacket(const NetEqInput::PacketData& packet,
+                         NetEq* neteq) override {
+    if (last_ssrc_ && packet.header.ssrc != *last_ssrc_) {
+      std::cout << "Changing streams from 0x" << std::hex << *last_ssrc_
+                << " to 0x" << std::hex << packet.header.ssrc << std::dec
+                << " (payload type "
+                << static_cast<int>(packet.header.payloadType) << ")"
+                << std::endl;
+    }
+    last_ssrc_ = packet.header.ssrc;
+    if (other_callback_) {
+      other_callback_->AfterInsertPacket(packet, neteq);
+    }
+  }
+
+ private:
+  NetEqPostInsertPacket* other_callback_;
+  absl::optional<uint32_t> last_ssrc_;
+};
+
+NetEqTestFactory::NetEqTestFactory() = default;
+NetEqTestFactory::~NetEqTestFactory() = default;
+
+NetEqTestFactory::Config::Config() = default;
+NetEqTestFactory::Config::Config(const Config& other) = default;
+NetEqTestFactory::Config::~Config() = default;
+
+std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTestFromString(
+    absl::string_view input_string,
+    NetEqFactory* factory,
+    const Config& config) {
+  std::unique_ptr<NetEqInput> input(
+      NetEqEventLogInput::CreateFromString(input_string, config.ssrc_filter));
+  if (!input) {
+    std::cerr << "Error: Cannot parse input string" << std::endl;
+    return nullptr;
+  }
+  return InitializeTest(std::move(input), factory, config);
+}
+
+std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTestFromFile(
+    absl::string_view input_file_name,
+    NetEqFactory* factory,
+    const Config& config) {
+  // Gather RTP header extensions in a map.
+  NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
+      {config.audio_level, kRtpExtensionAudioLevel},
+      {config.abs_send_time, kRtpExtensionAbsoluteSendTime},
+      {config.transport_seq_no, kRtpExtensionTransportSequenceNumber},
+      {config.video_content_type, kRtpExtensionVideoContentType},
+      {config.video_timing, kRtpExtensionVideoTiming}};
+
+  std::unique_ptr<NetEqInput> input;
+  if (RtpFileSource::ValidRtpDump(input_file_name) ||
+      RtpFileSource::ValidPcap(input_file_name)) {
+    input.reset(new NetEqRtpDumpInput(input_file_name, rtp_ext_map,
+                                      config.ssrc_filter));
+  } else {
+    input.reset(NetEqEventLogInput::CreateFromFile(input_file_name,
+                                                   config.ssrc_filter));
+  }
+
+  std::cout << "Input file: " << input_file_name << std::endl;
+  if (!input) {
+    std::cerr << "Error: Cannot open input file" << std::endl;
+    return nullptr;
+  }
+  return InitializeTest(std::move(input), factory, config);
+}
+
+std::unique_ptr<NetEqTest> NetEqTestFactory::InitializeTest(
+    std::unique_ptr<NetEqInput> input,
+    NetEqFactory* factory,
+    const Config& config) {
+  if (input->ended()) {
+    std::cerr << "Error: Input is empty" << std::endl;
+    return nullptr;
+  }
+
+  if (!config.field_trial_string.empty()) {
+    field_trials_ =
+        std::make_unique<ScopedFieldTrials>(config.field_trial_string);
+  }
+
+  // Skip some initial events/packets if requested.
+  if (config.skip_get_audio_events > 0) {
+    std::cout << "Skipping " << config.skip_get_audio_events
+              << " get_audio events" << std::endl;
+    if (!input->NextPacketTime() || !input->NextOutputEventTime()) {
+      std::cerr << "No events found" << std::endl;
+      return nullptr;
+    }
+    for (int i = 0; i < config.skip_get_audio_events; i++) {
+      input->AdvanceOutputEvent();
+      if (!input->NextOutputEventTime()) {
+        std::cerr << "Not enough get_audio events found" << std::endl;
+        return nullptr;
+      }
+    }
+    while (*input->NextPacketTime() < *input->NextOutputEventTime()) {
+      input->PopPacket();
+      if (!input->NextPacketTime()) {
+        std::cerr << "Not enough incoming packets found" << std::endl;
+        return nullptr;
+      }
+    }
+  }
+
+  // Check the sample rate.
+  absl::optional<int> sample_rate_hz;
+  std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
+  while (absl::optional<RTPHeader> first_rtp_header = input->NextHeader()) {
+    RTC_DCHECK(first_rtp_header);
+    sample_rate_hz = CodecSampleRate(first_rtp_header->payloadType, config);
+    if (sample_rate_hz) {
+      std::cout << "Found valid packet with payload type "
+                << static_cast<int>(first_rtp_header->payloadType)
+                << " and SSRC 0x" << std::hex << first_rtp_header->ssrc
+                << std::dec << std::endl;
+      if (config.initial_dummy_packets > 0) {
+        std::cout << "Nr of initial dummy packets: "
+                  << config.initial_dummy_packets << std::endl;
+        input = std::make_unique<InitialPacketInserterNetEqInput>(
+            std::move(input), config.initial_dummy_packets, *sample_rate_hz);
+      }
+      break;
+    }
+    // Discard this packet and move to the next. Keep track of discarded payload
+    // types and SSRCs.
+    discarded_pt_and_ssrc.emplace(first_rtp_header->payloadType,
+                                  first_rtp_header->ssrc);
+    input->PopPacket();
+  }
+  if (!discarded_pt_and_ssrc.empty()) {
+    std::cout << "Discarded initial packets with the following payload types "
+                 "and SSRCs:"
+              << std::endl;
+    for (const auto& d : discarded_pt_and_ssrc) {
+      std::cout << "PT " << d.first << "; SSRC 0x" << std::hex
+                << static_cast<int>(d.second) << std::dec << std::endl;
+    }
+  }
+  if (!sample_rate_hz) {
+    std::cerr << "Cannot find any packets with known payload types"
+              << std::endl;
+    return nullptr;
+  }
+
+  // If an output file is requested, open it.
+  std::unique_ptr<AudioSink> output;
+  if (!config.output_audio_filename.has_value()) {
+    output = std::make_unique<VoidAudioSink>();
+    std::cout << "No output audio file" << std::endl;
+  } else if (config.output_audio_filename->size() >= 4 &&
+             config.output_audio_filename->substr(
+                 config.output_audio_filename->size() - 4) == ".wav") {
+    // Open a wav file with the known sample rate.
+    output = std::make_unique<OutputWavFile>(*config.output_audio_filename,
+                                             *sample_rate_hz);
+    std::cout << "Output WAV file: " << *config.output_audio_filename
+              << std::endl;
+  } else {
+    // Open a pcm file.
+    output = std::make_unique<OutputAudioFile>(*config.output_audio_filename);
+    std::cout << "Output PCM file: " << *config.output_audio_filename
+              << std::endl;
+  }
+
+  NetEqTest::DecoderMap codecs = NetEqTest::StandardDecoderMap();
+
+  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory =
+      CreateBuiltinAudioDecoderFactory();
+
+  // Check if a replacement audio file was provided.
+  if (config.replacement_audio_file.size() > 0) {
+    // Find largest unused payload type.
+    int replacement_pt = 127;
+    while (codecs.find(replacement_pt) != codecs.end()) {
+      --replacement_pt;
+      if (replacement_pt <= 0) {
+        std::cerr << "Error: Unable to find available replacement payload type"
+                  << std::endl;
+        return nullptr;
+      }
+    }
+
+    auto std_set_int32_to_uint8 = [](const std::set<int32_t>& a) {
+      std::set<uint8_t> b;
+      for (auto& x : a) {
+        b.insert(static_cast<uint8_t>(x));
+      }
+      return b;
+    };
+
+    std::set<uint8_t> cn_types = std_set_int32_to_uint8(
+        {config.cn_nb, config.cn_wb, config.cn_swb32, config.cn_swb48});
+    std::set<uint8_t> forbidden_types =
+        std_set_int32_to_uint8({config.g722, config.red, config.avt,
+                                config.avt_16, config.avt_32, config.avt_48});
+    input.reset(new NetEqReplacementInput(std::move(input), replacement_pt,
+                                          cn_types, forbidden_types));
+
+    // Note that capture-by-copy implies that the lambda captures the value of
+    // decoder_factory before it's reassigned on the left-hand side.
+    decoder_factory = rtc::make_ref_counted<FunctionAudioDecoderFactory>(
+        [decoder_factory, config](
+            const SdpAudioFormat& format,
+            absl::optional<AudioCodecPairId> codec_pair_id) {
+          std::unique_ptr<AudioDecoder> decoder =
+              decoder_factory->MakeAudioDecoder(format, codec_pair_id);
+          if (!decoder && format.name == "replacement") {
+            decoder = std::make_unique<FakeDecodeFromFile>(
+                std::make_unique<InputAudioFile>(config.replacement_audio_file),
+                format.clockrate_hz, format.num_channels > 1);
+          }
+          return decoder;
+        });
+
+    if (!codecs
+             .insert({replacement_pt, SdpAudioFormat("replacement", 48000, 1)})
+             .second) {
+      std::cerr << "Error: Unable to insert replacement audio codec"
+                << std::endl;
+      return nullptr;
+    }
+  }
+
+  // Create a text log output stream if needed.
+  std::unique_ptr<std::ofstream> text_log;
+  if (config.textlog && config.textlog_filename.has_value()) {
+    // Write to file.
+    text_log = std::make_unique<std::ofstream>(*config.textlog_filename);
+  } else if (config.textlog) {
+    // Print to stdout.
+    text_log = std::make_unique<std::ofstream>();
+    text_log->basic_ios<char>::rdbuf(std::cout.rdbuf());
+  }
+
+  NetEqTest::Callbacks callbacks;
+  stats_plotter_ = std::make_unique<NetEqStatsPlotter>(
+      config.matlabplot, config.pythonplot, config.concealment_events,
+      config.plot_scripts_basename.value_or(""));
+
+  ssrc_switch_detector_.reset(
+      new SsrcSwitchDetector(stats_plotter_->stats_getter()->delay_analyzer()));
+  callbacks.post_insert_packet = ssrc_switch_detector_.get();
+  callbacks.get_audio_callback = stats_plotter_->stats_getter();
+  callbacks.simulation_ended_callback = stats_plotter_.get();
+  NetEq::Config neteq_config;
+  neteq_config.sample_rate_hz = *sample_rate_hz;
+  neteq_config.max_packets_in_buffer = config.max_nr_packets_in_buffer;
+  neteq_config.enable_fast_accelerate = config.enable_fast_accelerate;
+  return std::make_unique<NetEqTest>(
+      neteq_config, decoder_factory, codecs, std::move(text_log), factory,
+      std::move(input), std::move(output), callbacks);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.h
new file mode 100644
index 0000000000..96ce0b4334
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_test_factory.h
@@ -0,0 +1,172 @@
+/*
+ *  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_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "test/field_trial.h"
+
+namespace webrtc {
+namespace test {
+
+class SsrcSwitchDetector;
+class NetEqStatsGetter;
+class NetEqStatsPlotter;
+
+// Note that the NetEqTestFactory needs to be alive when the NetEqTest object is
+// used for a simulation.
+class NetEqTestFactory {
+ public:
+  NetEqTestFactory();
+  ~NetEqTestFactory();
+  struct Config {
+    Config();
+    Config(const Config& other);
+    ~Config();
+    // RTP payload type for PCM-u.
+    static constexpr int default_pcmu() { return 0; }
+    int pcmu = default_pcmu();
+    // RTP payload type for PCM-a.
+    static constexpr int default_pcma() { return 8; }
+    int pcma = default_pcma();
+    // RTP payload type for iLBC.
+    static constexpr int default_ilbc() { return 102; }
+    int ilbc = default_ilbc();
+    // RTP payload type for iSAC.
+    static constexpr int default_isac() { return 103; }
+    int isac = default_isac();
+    // RTP payload type for iSAC-swb (32 kHz).
+    static constexpr int default_isac_swb() { return 104; }
+    int isac_swb = default_isac_swb();
+    // RTP payload type for Opus.
+    static constexpr int default_opus() { return 111; }
+    int opus = default_opus();
+    // RTP payload type for PCM16b-nb (8 kHz).
+    static constexpr int default_pcm16b() { return 93; }
+    int pcm16b = default_pcm16b();
+    // RTP payload type for PCM16b-wb (16 kHz).
+    static constexpr int default_pcm16b_wb() { return 94; }
+    int pcm16b_wb = default_pcm16b_wb();
+    // RTP payload type for PCM16b-swb32 (32 kHz).
+    static constexpr int default_pcm16b_swb32() { return 95; }
+    int pcm16b_swb32 = default_pcm16b_swb32();
+    // RTP payload type for PCM16b-swb48 (48 kHz).
+    static constexpr int default_pcm16b_swb48() { return 96; }
+    int pcm16b_swb48 = default_pcm16b_swb48();
+    // RTP payload type for G.722.
+    static constexpr int default_g722() { return 9; }
+    int g722 = default_g722();
+    // RTP payload type for AVT/DTMF (8 kHz).
+    static constexpr int default_avt() { return 106; }
+    int avt = default_avt();
+    // RTP payload type for AVT/DTMF (16 kHz).
+    static constexpr int default_avt_16() { return 114; }
+    int avt_16 = default_avt_16();
+    // RTP payload type for AVT/DTMF (32 kHz).
+    static constexpr int default_avt_32() { return 115; }
+    int avt_32 = default_avt_32();
+    // RTP payload type for AVT/DTMF (48 kHz).
+    static constexpr int default_avt_48() { return 116; }
+    int avt_48 = default_avt_48();
+    // RTP payload type for redundant audio (RED).
+    static constexpr int default_red() { return 117; }
+    int red = default_red();
+    // RTP payload type for comfort noise (8 kHz).
+    static constexpr int default_cn_nb() { return 13; }
+    int cn_nb = default_cn_nb();
+    // RTP payload type for comfort noise (16 kHz).
+    static constexpr int default_cn_wb() { return 98; }
+    int cn_wb = default_cn_wb();
+    // RTP payload type for comfort noise (32 kHz).
+    static constexpr int default_cn_swb32() { return 99; }
+    int cn_swb32 = default_cn_swb32();
+    // RTP payload type for comfort noise (48 kHz).
+    static constexpr int default_cn_swb48() { return 100; }
+    int cn_swb48 = default_cn_swb48();
+    // A PCM file that will be used to populate dummy RTP packets.
+    std::string replacement_audio_file;
+    // Only use packets with this SSRC.
+    absl::optional<uint32_t> ssrc_filter;
+    // Extension ID for audio level (RFC 6464).
+    static constexpr int default_audio_level() { return 1; }
+    int audio_level = default_audio_level();
+    // Extension ID for absolute sender time.
+    static constexpr int default_abs_send_time() { return 3; }
+    int abs_send_time = default_abs_send_time();
+    // Extension ID for transport sequence number.
+    static constexpr int default_transport_seq_no() { return 5; }
+    int transport_seq_no = default_transport_seq_no();
+    // Extension ID for video content type.
+    static constexpr int default_video_content_type() { return 7; }
+    int video_content_type = default_video_content_type();
+    // Extension ID for video timing.
+    static constexpr int default_video_timing() { return 8; }
+    int video_timing = default_video_timing();
+    // Generate a matlab script for plotting the delay profile.
+    bool matlabplot = false;
+    // Generates a python script for plotting the delay profile.
+    bool pythonplot = false;
+    // Prints concealment events.
+    bool concealment_events = false;
+    // Maximum allowed number of packets in the buffer.
+    static constexpr int default_max_nr_packets_in_buffer() { return 200; }
+    int max_nr_packets_in_buffer = default_max_nr_packets_in_buffer();
+    // Number of dummy packets to put in the packet buffer at the start of the
+    // simulation.
+    static constexpr int default_initial_dummy_packets() { return 0; }
+    int initial_dummy_packets = default_initial_dummy_packets();
+    // Number of getAudio events to skip at the start of the simulation.
+    static constexpr int default_skip_get_audio_events() { return 0; }
+    int skip_get_audio_events = default_skip_get_audio_events();
+    // Enables jitter buffer fast accelerate.
+    bool enable_fast_accelerate = false;
+    // Dumps events that describes the simulation on a step-by-step basis.
+    bool textlog = false;
+    // If specified and `textlog` is true, the output of `textlog` is written to
+    // the specified file name.
+    absl::optional<std::string> textlog_filename;
+    // Base name for the output script files for plotting the delay profile.
+    absl::optional<std::string> plot_scripts_basename;
+    // Path to the output audio file.
+    absl::optional<std::string> output_audio_filename;
+    // Field trials to use during the simulation.
+    std::string field_trial_string;
+  };
+
+  std::unique_ptr<NetEqTest> InitializeTestFromFile(
+      absl::string_view input_filename,
+      NetEqFactory* neteq_factory,
+      const Config& config);
+  std::unique_ptr<NetEqTest> InitializeTestFromString(
+      absl::string_view input_string,
+      NetEqFactory* neteq_factory,
+      const Config& config);
+
+ private:
+  std::unique_ptr<NetEqTest> InitializeTest(std::unique_ptr<NetEqInput> input,
+                                            NetEqFactory* neteq_factory,
+                                            const Config& config);
+  std::unique_ptr<SsrcSwitchDetector> ssrc_switch_detector_;
+  std::unique_ptr<NetEqStatsPlotter> stats_plotter_;
+  // The field trials are stored in the test factory, because neteq_test is not
+  // in a testonly target, and therefore cannot use ScopedFieldTrials.
+  std::unique_ptr<ScopedFieldTrials> field_trials_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_audio_file.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_audio_file.h
new file mode 100644
index 0000000000..25577fc882
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_audio_file.h
@@ -0,0 +1,51 @@
+/*
+ *  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_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
+
+#include <stdio.h>
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+
+namespace webrtc {
+namespace test {
+
+class OutputAudioFile : public AudioSink {
+ public:
+  // Creates an OutputAudioFile, opening a file named `file_name` for writing.
+  // The file format is 16-bit signed host-endian PCM.
+  explicit OutputAudioFile(absl::string_view file_name) {
+    out_file_ = fopen(std::string(file_name).c_str(), "wb");
+  }
+
+  virtual ~OutputAudioFile() {
+    if (out_file_)
+      fclose(out_file_);
+  }
+
+  OutputAudioFile(const OutputAudioFile&) = delete;
+  OutputAudioFile& operator=(const OutputAudioFile&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
+    RTC_DCHECK(out_file_);
+    return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
+  }
+
+ private:
+  FILE* out_file_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_wav_file.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_wav_file.h
new file mode 100644
index 0000000000..20eedfb554
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/output_wav_file.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_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+
+namespace webrtc {
+namespace test {
+
+class OutputWavFile : public AudioSink {
+ public:
+  // Creates an OutputWavFile, opening a file named `file_name` for writing.
+  // The output file is a PCM encoded wav file.
+  OutputWavFile(absl::string_view file_name,
+                int sample_rate_hz,
+                int num_channels = 1)
+      : wav_writer_(file_name, sample_rate_hz, num_channels) {}
+
+  OutputWavFile(const OutputWavFile&) = delete;
+  OutputWavFile& operator=(const OutputWavFile&) = delete;
+
+  bool WriteArray(const int16_t* audio, size_t num_samples) override {
+    wav_writer_.WriteSamples(audio, num_samples);
+    return true;
+  }
+
+ private:
+  WavWriter wav_writer_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.cc
new file mode 100644
index 0000000000..e540173f43
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.cc
@@ -0,0 +1,133 @@
+/*
+ *  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/neteq/tools/packet.h"
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+namespace test {
+
+Packet::Packet(rtc::CopyOnWriteBuffer packet,
+               size_t virtual_packet_length_bytes,
+               double time_ms,
+               const RtpHeaderExtensionMap* extension_map)
+    : packet_(std::move(packet)),
+      virtual_packet_length_bytes_(virtual_packet_length_bytes),
+      time_ms_(time_ms),
+      valid_header_(ParseHeader(extension_map)) {}
+
+Packet::Packet(const RTPHeader& header,
+               size_t virtual_packet_length_bytes,
+               size_t virtual_payload_length_bytes,
+               double time_ms)
+    : header_(header),
+      virtual_packet_length_bytes_(virtual_packet_length_bytes),
+      virtual_payload_length_bytes_(virtual_payload_length_bytes),
+      time_ms_(time_ms),
+      valid_header_(true) {}
+
+Packet::~Packet() = default;
+
+bool Packet::ExtractRedHeaders(std::list<RTPHeader*>* headers) const {
+  //
+  //  0                   1                    2                   3
+  //  0 1 2 3 4 5 6 7 8 9 0 1 2 3  4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |1|   block PT  |  timestamp offset         |   block length    |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |1|    ...                                                      |
+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+  // |0|   block PT  |
+  // +-+-+-+-+-+-+-+-+
+  //
+
+  const uint8_t* payload_ptr = payload();
+  const uint8_t* payload_end_ptr = payload_ptr + payload_length_bytes();
+
+  // Find all RED headers with the extension bit set to 1. That is, all headers
+  // but the last one.
+  while ((payload_ptr < payload_end_ptr) && (*payload_ptr & 0x80)) {
+    RTPHeader* header = new RTPHeader;
+    CopyToHeader(header);
+    header->payloadType = payload_ptr[0] & 0x7F;
+    uint32_t offset = (payload_ptr[1] << 6) + ((payload_ptr[2] & 0xFC) >> 2);
+    header->timestamp -= offset;
+    headers->push_front(header);
+    payload_ptr += 4;
+  }
+  // Last header.
+  RTC_DCHECK_LT(payload_ptr, payload_end_ptr);
+  if (payload_ptr >= payload_end_ptr) {
+    return false;  // Payload too short.
+  }
+  RTPHeader* header = new RTPHeader;
+  CopyToHeader(header);
+  header->payloadType = payload_ptr[0] & 0x7F;
+  headers->push_front(header);
+  return true;
+}
+
+void Packet::DeleteRedHeaders(std::list<RTPHeader*>* headers) {
+  while (!headers->empty()) {
+    delete headers->front();
+    headers->pop_front();
+  }
+}
+
+bool Packet::ParseHeader(const RtpHeaderExtensionMap* extension_map) {
+  // Use RtpPacketReceived instead of RtpPacket because former already has a
+  // converter into legacy RTPHeader.
+  webrtc::RtpPacketReceived rtp_packet(extension_map);
+
+  // Because of the special case of dummy packets that have padding marked in
+  // the RTP header, but do not have rtp payload with the padding size, handle
+  // padding manually. Regular RTP packet parser reports failure, but it is fine
+  // in this context.
+  bool padding = (packet_[0] & 0b0010'0000);
+  size_t padding_size = 0;
+  if (padding) {
+    // Clear the padding bit to prevent failure when rtp payload is omited.
+    rtc::CopyOnWriteBuffer packet(packet_);
+    packet.MutableData()[0] &= ~0b0010'0000;
+    if (!rtp_packet.Parse(std::move(packet))) {
+      return false;
+    }
+    if (rtp_packet.payload_size() > 0) {
+      padding_size = rtp_packet.data()[rtp_packet.size() - 1];
+    }
+    if (padding_size > rtp_packet.payload_size()) {
+      return false;
+    }
+  } else {
+    if (!rtp_packet.Parse(packet_)) {
+      return false;
+    }
+  }
+  rtp_payload_ = rtc::MakeArrayView(packet_.data() + rtp_packet.headers_size(),
+                                    rtp_packet.payload_size() - padding_size);
+  rtp_packet.GetHeader(&header_);
+
+  RTC_CHECK_GE(virtual_packet_length_bytes_, rtp_packet.size());
+  RTC_DCHECK_GE(virtual_packet_length_bytes_, rtp_packet.headers_size());
+  virtual_payload_length_bytes_ =
+      virtual_packet_length_bytes_ - rtp_packet.headers_size();
+  return true;
+}
+
+void Packet::CopyToHeader(RTPHeader* destination) const {
+  *destination = header_;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.h
new file mode 100644
index 0000000000..96710907df
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet.h
@@ -0,0 +1,104 @@
+/*
+ *  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_NETEQ_TOOLS_PACKET_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
+
+#include <list>
+
+#include "api/array_view.h"
+#include "api/rtp_headers.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for handling RTP packets in test applications.
+class Packet {
+ public:
+  // Creates a packet, with the packet payload (including header bytes) in
+  // `packet`. The `time_ms` is an extra time associated with this packet,
+  // typically used to denote arrival time.
+  // `virtual_packet_length_bytes` is typically used when reading RTP dump files
+  // that only contain the RTP headers, and no payload (a.k.a RTP dummy files or
+  // RTP light). The `virtual_packet_length_bytes` tells what size the packet
+  // had on wire, including the now discarded payload.
+  Packet(rtc::CopyOnWriteBuffer packet,
+         size_t virtual_packet_length_bytes,
+         double time_ms,
+         const RtpHeaderExtensionMap* extension_map = nullptr);
+
+  Packet(rtc::CopyOnWriteBuffer packet,
+         double time_ms,
+         const RtpHeaderExtensionMap* extension_map = nullptr)
+      : Packet(packet, packet.size(), time_ms, extension_map) {}
+
+  // Same as above, but creates the packet from an already parsed RTPHeader.
+  // This is typically used when reading RTP dump files that only contain the
+  // RTP headers, and no payload. The `virtual_packet_length_bytes` tells what
+  // size the packet had on wire, including the now discarded payload,
+  // The `virtual_payload_length_bytes` tells the size of the payload.
+  Packet(const RTPHeader& header,
+         size_t virtual_packet_length_bytes,
+         size_t virtual_payload_length_bytes,
+         double time_ms);
+
+  virtual ~Packet();
+
+  Packet(const Packet&) = delete;
+  Packet& operator=(const Packet&) = delete;
+
+  // Parses the first bytes of the RTP payload, interpreting them as RED headers
+  // according to RFC 2198. The headers will be inserted into `headers`. The
+  // caller of the method assumes ownership of the objects in the list, and
+  // must delete them properly.
+  bool ExtractRedHeaders(std::list<RTPHeader*>* headers) const;
+
+  // Deletes all RTPHeader objects in `headers`, but does not delete `headers`
+  // itself.
+  static void DeleteRedHeaders(std::list<RTPHeader*>* headers);
+
+  const uint8_t* payload() const { return rtp_payload_.data(); }
+
+  size_t packet_length_bytes() const { return packet_.size(); }
+
+  size_t payload_length_bytes() const { return rtp_payload_.size(); }
+
+  size_t virtual_packet_length_bytes() const {
+    return virtual_packet_length_bytes_;
+  }
+
+  size_t virtual_payload_length_bytes() const {
+    return virtual_payload_length_bytes_;
+  }
+
+  const RTPHeader& header() const { return header_; }
+
+  double time_ms() const { return time_ms_; }
+  bool valid_header() const { return valid_header_; }
+
+ private:
+  bool ParseHeader(const RtpHeaderExtensionMap* extension_map);
+  void CopyToHeader(RTPHeader* destination) const;
+
+  RTPHeader header_;
+  const rtc::CopyOnWriteBuffer packet_;
+  rtc::ArrayView<const uint8_t> rtp_payload_;  // Empty for dummy RTP packets.
+  // Virtual lengths are used when parsing RTP header files (dummy RTP files).
+  const size_t virtual_packet_length_bytes_;
+  size_t virtual_payload_length_bytes_ = 0;
+  const double time_ms_;     // Used to denote a packet's arrival time.
+  const bool valid_header_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.cc
new file mode 100644
index 0000000000..598ae6edd4
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.cc
@@ -0,0 +1,25 @@
+/*
+ *  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/neteq/tools/packet_source.h"
+
+namespace webrtc {
+namespace test {
+
+PacketSource::PacketSource() = default;
+
+PacketSource::~PacketSource() = default;
+
+void PacketSource::FilterOutPayloadType(uint8_t payload_type) {
+  filter_.set(payload_type, true);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.h
new file mode 100644
index 0000000000..be1705cae1
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_source.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_NETEQ_TOOLS_PACKET_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
+
+#include <bitset>
+#include <memory>
+
+#include "modules/audio_coding/neteq/tools/packet.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for an object delivering RTP packets to test applications.
+class PacketSource {
+ public:
+  PacketSource();
+  virtual ~PacketSource();
+
+  PacketSource(const PacketSource&) = delete;
+  PacketSource& operator=(const PacketSource&) = delete;
+
+  // Returns next packet. Returns nullptr if the source is depleted, or if an
+  // error occurred.
+  virtual std::unique_ptr<Packet> NextPacket() = 0;
+
+  virtual void FilterOutPayloadType(uint8_t payload_type);
+
+ protected:
+  std::bitset<128> filter_;  // Payload type is 7 bits in the RFC.
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_unittest.cc
new file mode 100644
index 0000000000..69cf56b529
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/packet_unittest.cc
@@ -0,0 +1,226 @@
+/*
+ *  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.
+ */
+
+// Unit tests for test Packet class.
+
+#include "modules/audio_coding/neteq/tools/packet.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+const int kHeaderLengthBytes = 12;
+
+void MakeRtpHeader(int payload_type,
+                   int seq_number,
+                   uint32_t timestamp,
+                   uint32_t ssrc,
+                   uint8_t* rtp_data) {
+  rtp_data[0] = 0x80;
+  rtp_data[1] = static_cast<uint8_t>(payload_type);
+  rtp_data[2] = (seq_number >> 8) & 0xFF;
+  rtp_data[3] = (seq_number)&0xFF;
+  rtp_data[4] = timestamp >> 24;
+  rtp_data[5] = (timestamp >> 16) & 0xFF;
+  rtp_data[6] = (timestamp >> 8) & 0xFF;
+  rtp_data[7] = timestamp & 0xFF;
+  rtp_data[8] = ssrc >> 24;
+  rtp_data[9] = (ssrc >> 16) & 0xFF;
+  rtp_data[10] = (ssrc >> 8) & 0xFF;
+  rtp_data[11] = ssrc & 0xFF;
+}
+}  // namespace
+
+TEST(TestPacket, RegularPacket) {
+  const size_t kPacketLengthBytes = 100;
+  rtc::CopyOnWriteBuffer packet_memory(kPacketLengthBytes);
+  const uint8_t kPayloadType = 17;
+  const uint16_t kSequenceNumber = 4711;
+  const uint32_t kTimestamp = 47114711;
+  const uint32_t kSsrc = 0x12345678;
+  MakeRtpHeader(kPayloadType, kSequenceNumber, kTimestamp, kSsrc,
+                packet_memory.MutableData());
+  const double kPacketTime = 1.0;
+  Packet packet(std::move(packet_memory), kPacketTime);
+  ASSERT_TRUE(packet.valid_header());
+  EXPECT_EQ(kPayloadType, packet.header().payloadType);
+  EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber);
+  EXPECT_EQ(kTimestamp, packet.header().timestamp);
+  EXPECT_EQ(kSsrc, packet.header().ssrc);
+  EXPECT_EQ(0, packet.header().numCSRCs);
+  EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.payload_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes, packet.virtual_packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.virtual_payload_length_bytes());
+  EXPECT_EQ(kPacketTime, packet.time_ms());
+}
+
+TEST(TestPacket, DummyPacket) {
+  const size_t kPacketLengthBytes = kHeaderLengthBytes;  // Only RTP header.
+  const size_t kVirtualPacketLengthBytes = 100;
+  rtc::CopyOnWriteBuffer packet_memory(kPacketLengthBytes);
+  const uint8_t kPayloadType = 17;
+  const uint16_t kSequenceNumber = 4711;
+  const uint32_t kTimestamp = 47114711;
+  const uint32_t kSsrc = 0x12345678;
+  MakeRtpHeader(kPayloadType, kSequenceNumber, kTimestamp, kSsrc,
+                packet_memory.MutableData());
+  const double kPacketTime = 1.0;
+  Packet packet(std::move(packet_memory), kVirtualPacketLengthBytes,
+                kPacketTime);
+  ASSERT_TRUE(packet.valid_header());
+  EXPECT_EQ(kPayloadType, packet.header().payloadType);
+  EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber);
+  EXPECT_EQ(kTimestamp, packet.header().timestamp);
+  EXPECT_EQ(kSsrc, packet.header().ssrc);
+  EXPECT_EQ(0, packet.header().numCSRCs);
+  EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.payload_length_bytes());
+  EXPECT_EQ(kVirtualPacketLengthBytes, packet.virtual_packet_length_bytes());
+  EXPECT_EQ(kVirtualPacketLengthBytes - kHeaderLengthBytes,
+            packet.virtual_payload_length_bytes());
+  EXPECT_EQ(kPacketTime, packet.time_ms());
+}
+
+TEST(TestPacket, DummyPaddingPacket) {
+  const size_t kPacketLengthBytes = kHeaderLengthBytes;  // Only RTP header.
+  const size_t kVirtualPacketLengthBytes = 100;
+  rtc::CopyOnWriteBuffer packet_memory(kPacketLengthBytes);
+  const uint8_t kPayloadType = 17;
+  const uint16_t kSequenceNumber = 4711;
+  const uint32_t kTimestamp = 47114711;
+  const uint32_t kSsrc = 0x12345678;
+  MakeRtpHeader(kPayloadType, kSequenceNumber, kTimestamp, kSsrc,
+                packet_memory.MutableData());
+  packet_memory.MutableData()[0] |= 0b0010'0000;  // Set the padding bit.
+  const double kPacketTime = 1.0;
+  Packet packet(std::move(packet_memory), kVirtualPacketLengthBytes,
+                kPacketTime);
+  ASSERT_TRUE(packet.valid_header());
+  EXPECT_EQ(kPayloadType, packet.header().payloadType);
+  EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber);
+  EXPECT_EQ(kTimestamp, packet.header().timestamp);
+  EXPECT_EQ(kSsrc, packet.header().ssrc);
+  EXPECT_EQ(0, packet.header().numCSRCs);
+  EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.payload_length_bytes());
+  EXPECT_EQ(kVirtualPacketLengthBytes, packet.virtual_packet_length_bytes());
+  EXPECT_EQ(kVirtualPacketLengthBytes - kHeaderLengthBytes,
+            packet.virtual_payload_length_bytes());
+  EXPECT_EQ(kPacketTime, packet.time_ms());
+}
+
+namespace {
+// Writes one RED block header starting at `rtp_data`, according to RFC 2198.
+// returns the number of bytes written (1 or 4).
+//
+// Format if `last_payoad` is false:
+// 0                   1                    2                   3
+// 0 1 2 3 4 5 6 7 8 9 0 1 2 3  4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// |1|   block PT  |  timestamp offset         |   block length    |
+// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//
+// Format if `last_payoad` is true:
+// 0 1 2 3 4 5 6 7
+// +-+-+-+-+-+-+-+-+
+// |0|   Block PT  |
+// +-+-+-+-+-+-+-+-+
+
+int MakeRedHeader(int payload_type,
+                  uint32_t timestamp_offset,
+                  int block_length,
+                  bool last_payload,
+                  uint8_t* rtp_data) {
+  rtp_data[0] = 0x80 | (payload_type & 0x7F);  // Set the first bit to 1.
+  if (last_payload) {
+    rtp_data[0] &= 0x7F;  // Reset the first but to 0 to indicate last block.
+    return 1;
+  }
+  rtp_data[1] = timestamp_offset >> 6;
+  rtp_data[2] = (timestamp_offset & 0x3F) << 2;
+  rtp_data[2] |= block_length >> 8;
+  rtp_data[3] = block_length & 0xFF;
+  return 4;
+}
+}  // namespace
+
+TEST(TestPacket, RED) {
+  const size_t kPacketLengthBytes = 100;
+  rtc::CopyOnWriteBuffer packet_memory(kPacketLengthBytes);
+  const uint8_t kRedPayloadType = 17;
+  const uint16_t kSequenceNumber = 4711;
+  const uint32_t kTimestamp = 47114711;
+  const uint32_t kSsrc = 0x12345678;
+  MakeRtpHeader(kRedPayloadType, kSequenceNumber, kTimestamp, kSsrc,
+                packet_memory.MutableData());
+  // Create four RED headers.
+  // Payload types are just the same as the block index the offset is 100 times
+  // the block index.
+  const int kRedBlocks = 4;
+  uint8_t* payload_ptr = packet_memory.MutableData() +
+                         kHeaderLengthBytes;  // First byte after header.
+  for (int i = 0; i < kRedBlocks; ++i) {
+    int payload_type = i;
+    // Offset value is not used for the last block.
+    uint32_t timestamp_offset = 100 * i;
+    int block_length = 10 * i;
+    bool last_block = (i == kRedBlocks - 1) ? true : false;
+    payload_ptr += MakeRedHeader(payload_type, timestamp_offset, block_length,
+                                 last_block, payload_ptr);
+  }
+  const double kPacketTime = 1.0;
+  // Hand over ownership of `packet_memory` to `packet`.
+  Packet packet(packet_memory, kPacketLengthBytes, kPacketTime);
+  ASSERT_TRUE(packet.valid_header());
+  EXPECT_EQ(kRedPayloadType, packet.header().payloadType);
+  EXPECT_EQ(kSequenceNumber, packet.header().sequenceNumber);
+  EXPECT_EQ(kTimestamp, packet.header().timestamp);
+  EXPECT_EQ(kSsrc, packet.header().ssrc);
+  EXPECT_EQ(0, packet.header().numCSRCs);
+  EXPECT_EQ(kPacketLengthBytes, packet.packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.payload_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes, packet.virtual_packet_length_bytes());
+  EXPECT_EQ(kPacketLengthBytes - kHeaderLengthBytes,
+            packet.virtual_payload_length_bytes());
+  EXPECT_EQ(kPacketTime, packet.time_ms());
+  std::list<RTPHeader*> red_headers;
+  EXPECT_TRUE(packet.ExtractRedHeaders(&red_headers));
+  EXPECT_EQ(kRedBlocks, static_cast<int>(red_headers.size()));
+  int block_index = 0;
+  for (std::list<RTPHeader*>::reverse_iterator it = red_headers.rbegin();
+       it != red_headers.rend(); ++it) {
+    // Reading list from the back, since the extraction puts the main payload
+    // (which is the last one on wire) first.
+    RTPHeader* red_block = *it;
+    EXPECT_EQ(block_index, red_block->payloadType);
+    EXPECT_EQ(kSequenceNumber, red_block->sequenceNumber);
+    if (block_index == kRedBlocks - 1) {
+      // Last block has zero offset per definition.
+      EXPECT_EQ(kTimestamp, red_block->timestamp);
+    } else {
+      EXPECT_EQ(kTimestamp - 100 * block_index, red_block->timestamp);
+    }
+    EXPECT_EQ(kSsrc, red_block->ssrc);
+    EXPECT_EQ(0, red_block->numCSRCs);
+    ++block_index;
+  }
+  Packet::DeleteRedHeaders(&red_headers);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
new file mode 100644
index 0000000000..5050e1fb17
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
@@ -0,0 +1,48 @@
+/*
+ *  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/neteq/tools/resample_input_audio_file.h"
+
+#include <memory>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+bool ResampleInputAudioFile::Read(size_t samples,
+                                  int output_rate_hz,
+                                  int16_t* destination) {
+  const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
+  RTC_CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
+      << "Frame size and sample rates don't add up to an integer.";
+  std::unique_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
+  if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
+    return false;
+  resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
+  size_t output_length = 0;
+  RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read,
+                               destination, samples, output_length),
+               0);
+  RTC_CHECK_EQ(samples, output_length);
+  return true;
+}
+
+bool ResampleInputAudioFile::Read(size_t samples, int16_t* destination) {
+  RTC_CHECK_GT(output_rate_hz_, 0) << "Output rate not set.";
+  return Read(samples, output_rate_hz_, destination);
+}
+
+void ResampleInputAudioFile::set_output_rate_hz(int rate_hz) {
+  output_rate_hz_ = rate_hz;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
new file mode 100644
index 0000000000..279fece616
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
@@ -0,0 +1,55 @@
+/*
+ *  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_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/resampler/include/resampler.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for handling a looping input audio file with resampling.
+class ResampleInputAudioFile : public InputAudioFile {
+ public:
+  ResampleInputAudioFile(absl::string_view file_name,
+                         int file_rate_hz,
+                         bool loop_at_end = true)
+      : InputAudioFile(file_name, loop_at_end),
+        file_rate_hz_(file_rate_hz),
+        output_rate_hz_(-1) {}
+  ResampleInputAudioFile(absl::string_view file_name,
+                         int file_rate_hz,
+                         int output_rate_hz,
+                         bool loop_at_end = true)
+      : InputAudioFile(file_name, loop_at_end),
+        file_rate_hz_(file_rate_hz),
+        output_rate_hz_(output_rate_hz) {}
+
+  ResampleInputAudioFile(const ResampleInputAudioFile&) = delete;
+  ResampleInputAudioFile& operator=(const ResampleInputAudioFile&) = delete;
+
+  bool Read(size_t samples, int output_rate_hz, int16_t* destination);
+  bool Read(size_t samples, int16_t* destination) override;
+  void set_output_rate_hz(int rate_hz);
+
+ private:
+  const int file_rate_hz_;
+  int output_rate_hz_;
+  Resampler resampler_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
new file mode 100644
index 0000000000..1407aab5f2
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -0,0 +1,168 @@
+/*
+ *  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/neteq/tools/rtc_event_log_source.h"
+
+#include <string.h>
+
+#include <iostream>
+#include <limits>
+#include <memory>
+#include <set>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "logging/rtc_event_log/rtc_event_processor.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+bool ShouldSkipStream(ParsedRtcEventLog::MediaType media_type,
+                      uint32_t ssrc,
+                      absl::optional<uint32_t> ssrc_filter) {
+  if (media_type != ParsedRtcEventLog::MediaType::AUDIO)
+    return true;
+  if (ssrc_filter.has_value() && ssrc != *ssrc_filter)
+    return true;
+  return false;
+}
+}  // namespace
+
+std::unique_ptr<RtcEventLogSource> RtcEventLogSource::CreateFromFile(
+    absl::string_view file_name,
+    absl::optional<uint32_t> ssrc_filter) {
+  auto source = std::unique_ptr<RtcEventLogSource>(new RtcEventLogSource());
+  ParsedRtcEventLog parsed_log;
+  auto status = parsed_log.ParseFile(file_name);
+  if (!status.ok()) {
+    std::cerr << "Failed to parse event log: " << status.message() << std::endl;
+    std::cerr << "Skipping log." << std::endl;
+    return nullptr;
+  }
+  if (!source->Initialize(parsed_log, ssrc_filter)) {
+    std::cerr << "Failed to initialize source from event log, skipping."
+              << std::endl;
+    return nullptr;
+  }
+  return source;
+}
+
+std::unique_ptr<RtcEventLogSource> RtcEventLogSource::CreateFromString(
+    absl::string_view file_contents,
+    absl::optional<uint32_t> ssrc_filter) {
+  auto source = std::unique_ptr<RtcEventLogSource>(new RtcEventLogSource());
+  ParsedRtcEventLog parsed_log;
+  auto status = parsed_log.ParseString(file_contents);
+  if (!status.ok()) {
+    std::cerr << "Failed to parse event log: " << status.message() << std::endl;
+    std::cerr << "Skipping log." << std::endl;
+    return nullptr;
+  }
+  if (!source->Initialize(parsed_log, ssrc_filter)) {
+    std::cerr << "Failed to initialize source from event log, skipping."
+              << std::endl;
+    return nullptr;
+  }
+  return source;
+}
+
+RtcEventLogSource::~RtcEventLogSource() {}
+
+std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
+  if (rtp_packet_index_ >= rtp_packets_.size())
+    return nullptr;
+
+  std::unique_ptr<Packet> packet = std::move(rtp_packets_[rtp_packet_index_++]);
+  return packet;
+}
+
+int64_t RtcEventLogSource::NextAudioOutputEventMs() {
+  if (audio_output_index_ >= audio_outputs_.size())
+    return std::numeric_limits<int64_t>::max();
+
+  int64_t output_time_ms = audio_outputs_[audio_output_index_++];
+  return output_time_ms;
+}
+
+RtcEventLogSource::RtcEventLogSource() : PacketSource() {}
+
+bool RtcEventLogSource::Initialize(const ParsedRtcEventLog& parsed_log,
+                                   absl::optional<uint32_t> ssrc_filter) {
+  const auto first_log_end_time_us =
+      parsed_log.stop_log_events().empty()
+          ? std::numeric_limits<int64_t>::max()
+          : parsed_log.stop_log_events().front().log_time_us();
+
+  std::set<uint32_t> packet_ssrcs;
+  auto handle_rtp_packet =
+      [this, first_log_end_time_us,
+       &packet_ssrcs](const webrtc::LoggedRtpPacketIncoming& incoming) {
+        if (!filter_.test(incoming.rtp.header.payloadType) &&
+            incoming.log_time_us() < first_log_end_time_us) {
+          rtp_packets_.emplace_back(std::make_unique<Packet>(
+              incoming.rtp.header, incoming.rtp.total_length,
+              incoming.rtp.total_length - incoming.rtp.header_length,
+              static_cast<double>(incoming.log_time_ms())));
+          packet_ssrcs.insert(rtp_packets_.back()->header().ssrc);
+        }
+      };
+
+  std::set<uint32_t> ignored_ssrcs;
+  auto handle_audio_playout =
+      [this, first_log_end_time_us, &packet_ssrcs,
+       &ignored_ssrcs](const webrtc::LoggedAudioPlayoutEvent& audio_playout) {
+        if (audio_playout.log_time_us() < first_log_end_time_us) {
+          if (packet_ssrcs.count(audio_playout.ssrc) > 0) {
+            audio_outputs_.emplace_back(audio_playout.log_time_ms());
+          } else {
+            ignored_ssrcs.insert(audio_playout.ssrc);
+          }
+        }
+      };
+
+  // This wouldn't be needed if we knew that there was at most one audio stream.
+  webrtc::RtcEventProcessor event_processor;
+  for (const auto& rtp_packets : parsed_log.incoming_rtp_packets_by_ssrc()) {
+    ParsedRtcEventLog::MediaType media_type =
+        parsed_log.GetMediaType(rtp_packets.ssrc, webrtc::kIncomingPacket);
+    if (ShouldSkipStream(media_type, rtp_packets.ssrc, ssrc_filter)) {
+      continue;
+    }
+    event_processor.AddEvents(rtp_packets.incoming_packets, handle_rtp_packet);
+    // If no SSRC filter has been set, use the first SSRC only. The simulator
+    // does not work properly with interleaved packets from multiple SSRCs.
+    if (!ssrc_filter.has_value()) {
+      ssrc_filter = rtp_packets.ssrc;
+    }
+  }
+
+  for (const auto& audio_playouts : parsed_log.audio_playout_events()) {
+    if (ssrc_filter.has_value() && audio_playouts.first != *ssrc_filter)
+      continue;
+    event_processor.AddEvents(audio_playouts.second, handle_audio_playout);
+  }
+
+  // Fills in rtp_packets_ and audio_outputs_.
+  event_processor.ProcessEventsInOrder();
+
+  for (const auto& ssrc : ignored_ssrcs) {
+    std::cout << "Ignoring GetAudio events from SSRC 0x" << std::hex << ssrc
+              << " because no packets were found with a matching SSRC."
+              << std::endl;
+  }
+
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
new file mode 100644
index 0000000000..c67912a67d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -0,0 +1,71 @@
+/*
+ *  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_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class RtpHeaderParser;
+
+namespace test {
+
+class Packet;
+
+class RtcEventLogSource : public PacketSource {
+ public:
+  // Creates an RtcEventLogSource reading from `file_name`. If the file cannot
+  // be opened, or has the wrong format, NULL will be returned.
+  static std::unique_ptr<RtcEventLogSource> CreateFromFile(
+      absl::string_view file_name,
+      absl::optional<uint32_t> ssrc_filter);
+  // Same as above, but uses a string with the file contents.
+  static std::unique_ptr<RtcEventLogSource> CreateFromString(
+      absl::string_view file_contents,
+      absl::optional<uint32_t> ssrc_filter);
+
+  virtual ~RtcEventLogSource();
+
+  RtcEventLogSource(const RtcEventLogSource&) = delete;
+  RtcEventLogSource& operator=(const RtcEventLogSource&) = delete;
+
+  std::unique_ptr<Packet> NextPacket() override;
+
+  // Returns the timestamp of the next audio output event, in milliseconds. The
+  // maximum value of int64_t is returned if there are no more audio output
+  // events available.
+  int64_t NextAudioOutputEventMs();
+
+ private:
+  RtcEventLogSource();
+
+  bool Initialize(const ParsedRtcEventLog& parsed_log,
+                  absl::optional<uint32_t> ssrc_filter);
+
+  std::vector<std::unique_ptr<Packet>> rtp_packets_;
+  size_t rtp_packet_index_ = 0;
+  std::vector<int64_t> audio_outputs_;
+  size_t audio_output_index_ = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
new file mode 100644
index 0000000000..7ecf925ebb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -0,0 +1,166 @@
+/*
+ *  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 <stdio.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+
+ABSL_FLAG(int, red, 117, "RTP payload type for RED");
+ABSL_FLAG(int,
+          audio_level,
+          -1,
+          "Extension ID for audio level (RFC 6464); "
+          "-1 not to print audio level");
+ABSL_FLAG(int,
+          abs_send_time,
+          -1,
+          "Extension ID for absolute sender time; "
+          "-1 not to print absolute send time");
+
+int main(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  std::string usage =
+      "Tool for parsing an RTP dump file to text output.\n"
+      "Example usage:\n"
+      "./rtp_analyze input.rtp output.txt\n\n"
+      "Output is sent to stdout if no output file is given. "
+      "Note that this tool can read files with or without payloads.\n";
+  if (args.size() != 2 && args.size() != 3) {
+    printf("%s", usage.c_str());
+    return 1;
+  }
+
+  RTC_CHECK(absl::GetFlag(FLAGS_red) >= 0 &&
+            absl::GetFlag(FLAGS_red) <= 127);          // Payload type
+  RTC_CHECK(absl::GetFlag(FLAGS_audio_level) == -1 ||  // Default
+            (absl::GetFlag(FLAGS_audio_level) > 0 &&
+             absl::GetFlag(FLAGS_audio_level) <= 255));  // Extension ID
+  RTC_CHECK(absl::GetFlag(FLAGS_abs_send_time) == -1 ||  // Default
+            (absl::GetFlag(FLAGS_abs_send_time) > 0 &&
+             absl::GetFlag(FLAGS_abs_send_time) <= 255));  // Extension ID
+
+  printf("Input file: %s\n", args[1]);
+  std::unique_ptr<webrtc::test::RtpFileSource> file_source(
+      webrtc::test::RtpFileSource::Create(args[1]));
+  RTC_DCHECK(file_source.get());
+  // Set RTP extension IDs.
+  bool print_audio_level = false;
+  if (absl::GetFlag(FLAGS_audio_level) != -1) {
+    print_audio_level = true;
+    file_source->RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
+                                            absl::GetFlag(FLAGS_audio_level));
+  }
+  bool print_abs_send_time = false;
+  if (absl::GetFlag(FLAGS_abs_send_time) != -1) {
+    print_abs_send_time = true;
+    file_source->RegisterRtpHeaderExtension(
+        webrtc::kRtpExtensionAbsoluteSendTime,
+        absl::GetFlag(FLAGS_abs_send_time));
+  }
+
+  FILE* out_file;
+  if (args.size() == 3) {
+    out_file = fopen(args[2], "wt");
+    if (!out_file) {
+      printf("Cannot open output file %s\n", args[2]);
+      return -1;
+    }
+    printf("Output file: %s\n\n", args[2]);
+  } else {
+    out_file = stdout;
+  }
+
+  // Print file header.
+  fprintf(out_file, "SeqNo  TimeStamp   SendTime  Size    PT  M       SSRC");
+  if (print_audio_level) {
+    fprintf(out_file, " AuLvl (V)");
+  }
+  if (print_abs_send_time) {
+    fprintf(out_file, " AbsSendTime");
+  }
+  fprintf(out_file, "\n");
+
+  uint32_t max_abs_send_time = 0;
+  int cycles = -1;
+  std::unique_ptr<webrtc::test::Packet> packet;
+  while (true) {
+    packet = file_source->NextPacket();
+    if (!packet.get()) {
+      // End of file reached.
+      break;
+    }
+    // Write packet data to file. Use virtual_packet_length_bytes so that the
+    // correct packet sizes are printed also for RTP header-only dumps.
+    fprintf(out_file, "%5u %10u %10u %5i %5i %2i %#08X",
+            packet->header().sequenceNumber, packet->header().timestamp,
+            static_cast<unsigned int>(packet->time_ms()),
+            static_cast<int>(packet->virtual_packet_length_bytes()),
+            packet->header().payloadType, packet->header().markerBit,
+            packet->header().ssrc);
+    if (print_audio_level && packet->header().extension.hasAudioLevel) {
+      fprintf(out_file, " %5u (%1i)", packet->header().extension.audioLevel,
+              packet->header().extension.voiceActivity);
+    }
+    if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
+      if (cycles == -1) {
+        // Initialize.
+        max_abs_send_time = packet->header().extension.absoluteSendTime;
+        cycles = 0;
+      }
+      // Abs sender time is 24 bit 6.18 fixed point. Shift by 8 to normalize to
+      // 32 bits (unsigned). Calculate the difference between this packet's
+      // send time and the maximum observed. Cast to signed 32-bit to get the
+      // desired wrap-around behavior.
+      if (static_cast<int32_t>(
+              (packet->header().extension.absoluteSendTime << 8) -
+              (max_abs_send_time << 8)) >= 0) {
+        // The difference is non-negative, meaning that this packet is newer
+        // than the previously observed maximum absolute send time.
+        if (packet->header().extension.absoluteSendTime < max_abs_send_time) {
+          // Wrap detected.
+          cycles++;
+        }
+        max_abs_send_time = packet->header().extension.absoluteSendTime;
+      }
+      // Abs sender time is 24 bit 6.18 fixed point. Divide by 2^18 to convert
+      // to floating point representation.
+      double send_time_seconds =
+          static_cast<double>(packet->header().extension.absoluteSendTime) /
+              262144 +
+          64.0 * cycles;
+      fprintf(out_file, " %11f", send_time_seconds);
+    }
+    fprintf(out_file, "\n");
+
+    if (packet->header().payloadType == absl::GetFlag(FLAGS_red)) {
+      std::list<webrtc::RTPHeader*> red_headers;
+      packet->ExtractRedHeaders(&red_headers);
+      while (!red_headers.empty()) {
+        webrtc::RTPHeader* red = red_headers.front();
+        RTC_DCHECK(red);
+        fprintf(out_file, "* %5u %10u %10u %5i\n", red->sequenceNumber,
+                red->timestamp, static_cast<unsigned int>(packet->time_ms()),
+                red->payloadType);
+        red_headers.pop_front();
+        delete red;
+      }
+    }
+  }
+
+  fclose(out_file);
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_encode.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_encode.cc
new file mode 100644
index 0000000000..8adca927f0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_encode.cc
@@ -0,0 +1,351 @@
+/*
+ *  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 <stdio.h>
+
+#ifdef WIN32
+#include <winsock2.h>
+#endif
+#if defined(WEBRTC_LINUX) || defined(WEBRTC_FUCHSIA)
+#include <netinet/in.h>
+#endif
+
+#include <iostream>
+#include <map>
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "absl/memory/memory.h"
+#include "api/audio/audio_frame.h"
+#include "api/audio_codecs/L16/audio_encoder_L16.h"
+#include "api/audio_codecs/g711/audio_encoder_g711.h"
+#include "api/audio_codecs/g722/audio_encoder_g722.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+ABSL_FLAG(bool, list_codecs, false, "Enumerate all codecs");
+ABSL_FLAG(std::string, codec, "opus", "Codec to use");
+ABSL_FLAG(int,
+          frame_len,
+          0,
+          "Frame length in ms; 0 indicates codec default value");
+ABSL_FLAG(int, bitrate, 0, "Bitrate in kbps; 0 indicates codec default value");
+ABSL_FLAG(int,
+          payload_type,
+          -1,
+          "RTP payload type; -1 indicates codec default value");
+ABSL_FLAG(int,
+          cng_payload_type,
+          -1,
+          "RTP payload type for CNG; -1 indicates default value");
+ABSL_FLAG(int, ssrc, 0, "SSRC to write to the RTP header");
+ABSL_FLAG(bool, dtx, false, "Use DTX/CNG");
+ABSL_FLAG(int, sample_rate, 48000, "Sample rate of the input file");
+
+namespace webrtc {
+namespace test {
+namespace {
+
+// Add new codecs here, and to the map below.
+enum class CodecType {
+  kOpus,
+  kPcmU,
+  kPcmA,
+  kG722,
+  kPcm16b8,
+  kPcm16b16,
+  kPcm16b32,
+  kPcm16b48,
+  kIlbc,
+};
+
+struct CodecTypeAndInfo {
+  CodecType type;
+  int default_payload_type;
+  bool internal_dtx;
+};
+
+// List all supported codecs here. This map defines the command-line parameter
+// value (the key string) for selecting each codec, together with information
+// whether it is using internal or external DTX/CNG.
+const std::map<std::string, CodecTypeAndInfo>& CodecList() {
+  static const auto* const codec_list =
+      new std::map<std::string, CodecTypeAndInfo>{
+          {"opus", {CodecType::kOpus, 111, true}},
+          {"pcmu", {CodecType::kPcmU, 0, false}},
+          {"pcma", {CodecType::kPcmA, 8, false}},
+          {"g722", {CodecType::kG722, 9, false}},
+          {"pcm16b_8", {CodecType::kPcm16b8, 93, false}},
+          {"pcm16b_16", {CodecType::kPcm16b16, 94, false}},
+          {"pcm16b_32", {CodecType::kPcm16b32, 95, false}},
+          {"pcm16b_48", {CodecType::kPcm16b48, 96, false}},
+          {"ilbc", {CodecType::kIlbc, 102, false}}};
+  return *codec_list;
+}
+
+// This class will receive callbacks from ACM when a packet is ready, and write
+// it to the output file.
+class Packetizer : public AudioPacketizationCallback {
+ public:
+  Packetizer(FILE* out_file, uint32_t ssrc, int timestamp_rate_hz)
+      : out_file_(out_file),
+        ssrc_(ssrc),
+        timestamp_rate_hz_(timestamp_rate_hz) {}
+
+  int32_t SendData(AudioFrameType frame_type,
+                   uint8_t payload_type,
+                   uint32_t timestamp,
+                   const uint8_t* payload_data,
+                   size_t payload_len_bytes,
+                   int64_t absolute_capture_timestamp_ms) override {
+    if (payload_len_bytes == 0) {
+      return 0;
+    }
+
+    constexpr size_t kRtpHeaderLength = 12;
+    constexpr size_t kRtpDumpHeaderLength = 8;
+    const uint16_t length = htons(rtc::checked_cast<uint16_t>(
+        kRtpHeaderLength + kRtpDumpHeaderLength + payload_len_bytes));
+    const uint16_t plen = htons(
+        rtc::checked_cast<uint16_t>(kRtpHeaderLength + payload_len_bytes));
+    const uint32_t offset = htonl(timestamp / (timestamp_rate_hz_ / 1000));
+    RTC_CHECK_EQ(fwrite(&length, sizeof(uint16_t), 1, out_file_), 1);
+    RTC_CHECK_EQ(fwrite(&plen, sizeof(uint16_t), 1, out_file_), 1);
+    RTC_CHECK_EQ(fwrite(&offset, sizeof(uint32_t), 1, out_file_), 1);
+
+    const uint8_t rtp_header[] = {0x80,
+                                  static_cast<uint8_t>(payload_type & 0x7F),
+                                  static_cast<uint8_t>(sequence_number_ >> 8),
+                                  static_cast<uint8_t>(sequence_number_),
+                                  static_cast<uint8_t>(timestamp >> 24),
+                                  static_cast<uint8_t>(timestamp >> 16),
+                                  static_cast<uint8_t>(timestamp >> 8),
+                                  static_cast<uint8_t>(timestamp),
+                                  static_cast<uint8_t>(ssrc_ >> 24),
+                                  static_cast<uint8_t>(ssrc_ >> 16),
+                                  static_cast<uint8_t>(ssrc_ >> 8),
+                                  static_cast<uint8_t>(ssrc_)};
+    static_assert(sizeof(rtp_header) == kRtpHeaderLength, "");
+    RTC_CHECK_EQ(
+        fwrite(rtp_header, sizeof(uint8_t), kRtpHeaderLength, out_file_),
+        kRtpHeaderLength);
+    ++sequence_number_;  // Intended to wrap on overflow.
+
+    RTC_CHECK_EQ(
+        fwrite(payload_data, sizeof(uint8_t), payload_len_bytes, out_file_),
+        payload_len_bytes);
+
+    return 0;
+  }
+
+ private:
+  FILE* const out_file_;
+  const uint32_t ssrc_;
+  const int timestamp_rate_hz_;
+  uint16_t sequence_number_ = 0;
+};
+
+void SetFrameLenIfFlagIsPositive(int* config_frame_len) {
+  if (absl::GetFlag(FLAGS_frame_len) > 0) {
+    *config_frame_len = absl::GetFlag(FLAGS_frame_len);
+  }
+}
+
+template <typename T>
+typename T::Config GetCodecConfig() {
+  typename T::Config config;
+  SetFrameLenIfFlagIsPositive(&config.frame_size_ms);
+  RTC_CHECK(config.IsOk());
+  return config;
+}
+
+AudioEncoderL16::Config Pcm16bConfig(CodecType codec_type) {
+  auto config = GetCodecConfig<AudioEncoderL16>();
+  switch (codec_type) {
+    case CodecType::kPcm16b8:
+      config.sample_rate_hz = 8000;
+      return config;
+    case CodecType::kPcm16b16:
+      config.sample_rate_hz = 16000;
+      return config;
+    case CodecType::kPcm16b32:
+      config.sample_rate_hz = 32000;
+      return config;
+    case CodecType::kPcm16b48:
+      config.sample_rate_hz = 48000;
+      return config;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return config;
+  }
+}
+
+std::unique_ptr<AudioEncoder> CreateEncoder(CodecType codec_type,
+                                            int payload_type) {
+  switch (codec_type) {
+    case CodecType::kOpus: {
+      AudioEncoderOpus::Config config = GetCodecConfig<AudioEncoderOpus>();
+      if (absl::GetFlag(FLAGS_bitrate) > 0) {
+        config.bitrate_bps = absl::GetFlag(FLAGS_bitrate);
+      }
+      config.dtx_enabled = absl::GetFlag(FLAGS_dtx);
+      RTC_CHECK(config.IsOk());
+      return AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+    }
+
+    case CodecType::kPcmU:
+    case CodecType::kPcmA: {
+      AudioEncoderG711::Config config = GetCodecConfig<AudioEncoderG711>();
+      config.type = codec_type == CodecType::kPcmU
+                        ? AudioEncoderG711::Config::Type::kPcmU
+                        : AudioEncoderG711::Config::Type::kPcmA;
+      RTC_CHECK(config.IsOk());
+      return AudioEncoderG711::MakeAudioEncoder(config, payload_type);
+    }
+
+    case CodecType::kG722: {
+      return AudioEncoderG722::MakeAudioEncoder(
+          GetCodecConfig<AudioEncoderG722>(), payload_type);
+    }
+
+    case CodecType::kPcm16b8:
+    case CodecType::kPcm16b16:
+    case CodecType::kPcm16b32:
+    case CodecType::kPcm16b48: {
+      return AudioEncoderL16::MakeAudioEncoder(Pcm16bConfig(codec_type),
+                                               payload_type);
+    }
+
+    case CodecType::kIlbc: {
+      return AudioEncoderIlbc::MakeAudioEncoder(
+          GetCodecConfig<AudioEncoderIlbc>(), payload_type);
+    }
+  }
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+}
+
+AudioEncoderCngConfig GetCngConfig(int sample_rate_hz) {
+  AudioEncoderCngConfig cng_config;
+  const auto default_payload_type = [&] {
+    switch (sample_rate_hz) {
+      case 8000:
+        return 13;
+      case 16000:
+        return 98;
+      case 32000:
+        return 99;
+      case 48000:
+        return 100;
+      default:
+        RTC_DCHECK_NOTREACHED();
+    }
+    return 0;
+  };
+  cng_config.payload_type = absl::GetFlag(FLAGS_cng_payload_type) != -1
+                                ? absl::GetFlag(FLAGS_cng_payload_type)
+                                : default_payload_type();
+  return cng_config;
+}
+
+int RunRtpEncode(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  const std::string usage =
+      "Tool for generating an RTP dump file from audio input.\n"
+      "Example usage:\n"
+      "./rtp_encode input.pcm output.rtp --codec=[codec] "
+      "--frame_len=[frame_len] --bitrate=[bitrate]\n\n";
+  if (!absl::GetFlag(FLAGS_list_codecs) && args.size() != 3) {
+    printf("%s", usage.c_str());
+    return 1;
+  }
+
+  if (absl::GetFlag(FLAGS_list_codecs)) {
+    printf("The following arguments are valid --codec parameters:\n");
+    for (const auto& c : CodecList()) {
+      printf("  %s\n", c.first.c_str());
+    }
+    return 0;
+  }
+
+  const auto codec_it = CodecList().find(absl::GetFlag(FLAGS_codec));
+  if (codec_it == CodecList().end()) {
+    printf("%s is not a valid codec name.\n",
+           absl::GetFlag(FLAGS_codec).c_str());
+    printf("Use argument --list_codecs to see all valid codec names.\n");
+    return 1;
+  }
+
+  // Create the codec.
+  const int payload_type = absl::GetFlag(FLAGS_payload_type) == -1
+                               ? codec_it->second.default_payload_type
+                               : absl::GetFlag(FLAGS_payload_type);
+  std::unique_ptr<AudioEncoder> codec =
+      CreateEncoder(codec_it->second.type, payload_type);
+
+  // Create an external VAD/CNG encoder if needed.
+  if (absl::GetFlag(FLAGS_dtx) && !codec_it->second.internal_dtx) {
+    AudioEncoderCngConfig cng_config = GetCngConfig(codec->SampleRateHz());
+    RTC_DCHECK(codec);
+    cng_config.speech_encoder = std::move(codec);
+    codec = CreateComfortNoiseEncoder(std::move(cng_config));
+  }
+  RTC_DCHECK(codec);
+
+  // Set up ACM.
+  const int timestamp_rate_hz = codec->RtpTimestampRateHz();
+  AudioCodingModule::Config config;
+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(config));
+  acm->SetEncoder(std::move(codec));
+
+  // Open files.
+  printf("Input file: %s\n", args[1]);
+  InputAudioFile input_file(args[1], false);  // Open input in non-looping mode.
+  FILE* out_file = fopen(args[2], "wb");
+  RTC_CHECK(out_file) << "Could not open file " << args[2] << " for writing";
+  printf("Output file: %s\n", args[2]);
+  fprintf(out_file, "#!rtpplay1.0 \n");  //,
+  // Write 3 32-bit values followed by 2 16-bit values, all set to 0. This means
+  // a total of 16 bytes.
+  const uint8_t file_header[16] = {0};
+  RTC_CHECK_EQ(fwrite(file_header, sizeof(file_header), 1, out_file), 1);
+
+  // Create and register the packetizer, which will write the packets to file.
+  Packetizer packetizer(out_file, absl::GetFlag(FLAGS_ssrc), timestamp_rate_hz);
+  RTC_DCHECK_EQ(acm->RegisterTransportCallback(&packetizer), 0);
+
+  AudioFrame audio_frame;
+  audio_frame.samples_per_channel_ =
+      absl::GetFlag(FLAGS_sample_rate) / 100;  // 10 ms
+  audio_frame.sample_rate_hz_ = absl::GetFlag(FLAGS_sample_rate);
+  audio_frame.num_channels_ = 1;
+
+  while (input_file.Read(audio_frame.samples_per_channel_,
+                         audio_frame.mutable_data())) {
+    RTC_CHECK_GE(acm->Add10MsData(audio_frame), 0);
+    audio_frame.timestamp_ += audio_frame.samples_per_channel_;
+  }
+
+  return 0;
+}
+
+}  // namespace
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::RunRtpEncode(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
new file mode 100644
index 0000000000..a43c29638c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.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/neteq/tools/rtp_file_source.h"
+
+#include <string.h>
+
+#include "absl/strings/string_view.h"
+#ifndef WIN32
+#include <netinet/in.h>
+#endif
+
+#include <memory>
+
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
+#include "test/rtp_file_reader.h"
+
+namespace webrtc {
+namespace test {
+
+RtpFileSource* RtpFileSource::Create(absl::string_view file_name,
+                                     absl::optional<uint32_t> ssrc_filter) {
+  RtpFileSource* source = new RtpFileSource(ssrc_filter);
+  RTC_CHECK(source->OpenFile(file_name));
+  return source;
+}
+
+bool RtpFileSource::ValidRtpDump(absl::string_view file_name) {
+  std::unique_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
+  return !!temp_file;
+}
+
+bool RtpFileSource::ValidPcap(absl::string_view file_name) {
+  std::unique_ptr<RtpFileReader> temp_file(
+      RtpFileReader::Create(RtpFileReader::kPcap, file_name));
+  return !!temp_file;
+}
+
+RtpFileSource::~RtpFileSource() {}
+
+bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
+                                               uint8_t id) {
+  return rtp_header_extension_map_.RegisterByType(id, type);
+}
+
+std::unique_ptr<Packet> RtpFileSource::NextPacket() {
+  while (true) {
+    RtpPacket temp_packet;
+    if (!rtp_reader_->NextPacket(&temp_packet)) {
+      return NULL;
+    }
+    if (temp_packet.original_length == 0) {
+      // May be an RTCP packet.
+      // Read the next one.
+      continue;
+    }
+    auto packet = std::make_unique<Packet>(
+        rtc::CopyOnWriteBuffer(temp_packet.data, temp_packet.length),
+        temp_packet.original_length, temp_packet.time_ms,
+        &rtp_header_extension_map_);
+    if (!packet->valid_header()) {
+      continue;
+    }
+    if (filter_.test(packet->header().payloadType) ||
+        (ssrc_filter_ && packet->header().ssrc != *ssrc_filter_)) {
+      // This payload type should be filtered out. Continue to the next packet.
+      continue;
+    }
+    return packet;
+  }
+}
+
+RtpFileSource::RtpFileSource(absl::optional<uint32_t> ssrc_filter)
+    : PacketSource(),
+      ssrc_filter_(ssrc_filter) {}
+
+bool RtpFileSource::OpenFile(absl::string_view file_name) {
+  rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
+  if (rtp_reader_)
+    return true;
+  rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
+  if (!rtp_reader_) {
+    RTC_FATAL()
+        << "Couldn't open input file as either a rtpdump or .pcap. Note "
+        << "that .pcapng is not supported.";
+  }
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
new file mode 100644
index 0000000000..55505be630
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -0,0 +1,68 @@
+/*
+ *  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_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+namespace test {
+
+class RtpFileReader;
+
+class RtpFileSource : public PacketSource {
+ public:
+  // Creates an RtpFileSource reading from `file_name`. If the file cannot be
+  // opened, or has the wrong format, NULL will be returned.
+  static RtpFileSource* Create(
+      absl::string_view file_name,
+      absl::optional<uint32_t> ssrc_filter = absl::nullopt);
+
+  // Checks whether a files is a valid RTP dump or PCAP (Wireshark) file.
+  static bool ValidRtpDump(absl::string_view file_name);
+  static bool ValidPcap(absl::string_view file_name);
+
+  ~RtpFileSource() override;
+
+  RtpFileSource(const RtpFileSource&) = delete;
+  RtpFileSource& operator=(const RtpFileSource&) = delete;
+
+  // Registers an RTP header extension and binds it to `id`.
+  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
+
+  std::unique_ptr<Packet> NextPacket() override;
+
+ private:
+  static const int kFirstLineLength = 40;
+  static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
+  static const size_t kPacketHeaderSize = 8;
+
+  explicit RtpFileSource(absl::optional<uint32_t> ssrc_filter);
+
+  bool OpenFile(absl::string_view file_name);
+
+  std::unique_ptr<RtpFileReader> rtp_reader_;
+  const absl::optional<uint32_t> ssrc_filter_;
+  RtpHeaderExtensionMap rtp_header_extension_map_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc
new file mode 100644
index 0000000000..e883fc11d6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.cc
@@ -0,0 +1,60 @@
+/*
+ *  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 "modules/audio_coding/neteq/tools/rtp_generator.h"
+
+
+namespace webrtc {
+namespace test {
+
+uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
+                                    size_t payload_length_samples,
+                                    RTPHeader* rtp_header) {
+  RTC_DCHECK(rtp_header);
+  if (!rtp_header) {
+    return 0;
+  }
+  rtp_header->sequenceNumber = seq_number_++;
+  rtp_header->timestamp = timestamp_;
+  timestamp_ += static_cast<uint32_t>(payload_length_samples);
+  rtp_header->payloadType = payload_type;
+  rtp_header->markerBit = false;
+  rtp_header->ssrc = ssrc_;
+  rtp_header->numCSRCs = 0;
+
+  uint32_t this_send_time = next_send_time_ms_;
+  RTC_DCHECK_GT(samples_per_ms_, 0);
+  next_send_time_ms_ +=
+      ((1.0 + drift_factor_) * payload_length_samples) / samples_per_ms_;
+  return this_send_time;
+}
+
+void RtpGenerator::set_drift_factor(double factor) {
+  if (factor > -1.0) {
+    drift_factor_ = factor;
+  }
+}
+
+uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type,
+                                                 size_t payload_length_samples,
+                                                 RTPHeader* rtp_header) {
+  uint32_t ret = RtpGenerator::GetRtpHeader(payload_type,
+                                            payload_length_samples, rtp_header);
+  if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <=
+          jump_from_timestamp_ &&
+      timestamp_ > jump_from_timestamp_) {
+    // We just moved across the `jump_from_timestamp_` timestamp. Do the jump.
+    timestamp_ = jump_to_timestamp_;
+  }
+  return ret;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.h b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.h
new file mode 100644
index 0000000000..2e615adec5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -0,0 +1,83 @@
+/*
+ *  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_NETEQ_TOOLS_RTP_GENERATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+
+#include "api/rtp_headers.h"
+
+namespace webrtc {
+namespace test {
+
+// Class for generating RTP headers.
+class RtpGenerator {
+ public:
+  RtpGenerator(int samples_per_ms,
+               uint16_t start_seq_number = 0,
+               uint32_t start_timestamp = 0,
+               uint32_t start_send_time_ms = 0,
+               uint32_t ssrc = 0x12345678)
+      : seq_number_(start_seq_number),
+        timestamp_(start_timestamp),
+        next_send_time_ms_(start_send_time_ms),
+        ssrc_(ssrc),
+        samples_per_ms_(samples_per_ms),
+        drift_factor_(0.0) {}
+
+  virtual ~RtpGenerator() {}
+
+  RtpGenerator(const RtpGenerator&) = delete;
+  RtpGenerator& operator=(const RtpGenerator&) = delete;
+
+  // Writes the next RTP header to `rtp_header`, which will be of type
+  // `payload_type`. Returns the send time for this packet (in ms). The value of
+  // `payload_length_samples` determines the send time for the next packet.
+  virtual uint32_t GetRtpHeader(uint8_t payload_type,
+                                size_t payload_length_samples,
+                                RTPHeader* rtp_header);
+
+  void set_drift_factor(double factor);
+
+ protected:
+  uint16_t seq_number_;
+  uint32_t timestamp_;
+  uint32_t next_send_time_ms_;
+  const uint32_t ssrc_;
+  const int samples_per_ms_;
+  double drift_factor_;
+};
+
+class TimestampJumpRtpGenerator : public RtpGenerator {
+ public:
+  TimestampJumpRtpGenerator(int samples_per_ms,
+                            uint16_t start_seq_number,
+                            uint32_t start_timestamp,
+                            uint32_t jump_from_timestamp,
+                            uint32_t jump_to_timestamp)
+      : RtpGenerator(samples_per_ms, start_seq_number, start_timestamp),
+        jump_from_timestamp_(jump_from_timestamp),
+        jump_to_timestamp_(jump_to_timestamp) {}
+
+  TimestampJumpRtpGenerator(const TimestampJumpRtpGenerator&) = delete;
+  TimestampJumpRtpGenerator& operator=(const TimestampJumpRtpGenerator&) =
+      delete;
+
+  uint32_t GetRtpHeader(uint8_t payload_type,
+                        size_t payload_length_samples,
+                        RTPHeader* rtp_header) override;
+
+ private:
+  uint32_t jump_from_timestamp_;
+  uint32_t jump_to_timestamp_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc
new file mode 100644
index 0000000000..cccaa9a3bb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc
@@ -0,0 +1,148 @@
+/*
+ *  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 <stdio.h>
+
+#include <algorithm>
+#include <fstream>
+#include <iostream>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr size_t kRtpDumpHeaderLength = 8;
+
+// Returns the next packet or an empty buffer if end of file was encountered.
+rtc::Buffer ReadNextPacket(FILE* file) {
+  // Read the rtpdump header for the next packet.
+  rtc::Buffer buffer;
+  buffer.SetData(kRtpDumpHeaderLength, [&](rtc::ArrayView<uint8_t> x) {
+    return fread(x.data(), 1, x.size(), file);
+  });
+  if (buffer.size() != kRtpDumpHeaderLength) {
+    return rtc::Buffer();
+  }
+
+  // Get length field. This is the total length for this packet written to file,
+  // including the kRtpDumpHeaderLength bytes already read.
+  const uint16_t len = ByteReader<uint16_t>::ReadBigEndian(buffer.data());
+  RTC_CHECK_GE(len, kRtpDumpHeaderLength);
+
+  // Read remaining data from file directly into buffer.
+  buffer.AppendData(len - kRtpDumpHeaderLength, [&](rtc::ArrayView<uint8_t> x) {
+    return fread(x.data(), 1, x.size(), file);
+  });
+  if (buffer.size() != len) {
+    buffer.Clear();
+  }
+  return buffer;
+}
+
+struct PacketAndTime {
+  rtc::Buffer packet;
+  int time;
+};
+
+void WritePacket(const PacketAndTime& packet, FILE* file) {
+  // Write the first 4 bytes from the original packet.
+  const auto* payload_ptr = packet.packet.data();
+  RTC_CHECK_EQ(fwrite(payload_ptr, 4, 1, file), 1);
+  payload_ptr += 4;
+
+  // Convert the new time offset to network endian, and write to file.
+  uint8_t time[sizeof(uint32_t)];
+  ByteWriter<uint32_t, sizeof(uint32_t)>::WriteBigEndian(time, packet.time);
+  RTC_CHECK_EQ(fwrite(time, sizeof(uint32_t), 1, file), 1);
+  payload_ptr += 4;  // Skip the old time in the original payload.
+
+  // Write the remaining part of the payload.
+  RTC_DCHECK_EQ(payload_ptr - packet.packet.data(), kRtpDumpHeaderLength);
+  RTC_CHECK_EQ(
+      fwrite(payload_ptr, packet.packet.size() - kRtpDumpHeaderLength, 1, file),
+      1);
+}
+
+int RunRtpJitter(int argc, char* argv[]) {
+  const std::string program_name = argv[0];
+  const std::string usage =
+      "Tool for alternating the arrival times in an RTP dump file.\n"
+      "Example usage:\n" +
+      program_name + " input.rtp arrival_times_ms.txt output.rtp\n\n";
+  if (argc != 4) {
+    printf("%s", usage.c_str());
+    return 1;
+  }
+
+  printf("Input RTP file: %s\n", argv[1]);
+  FILE* in_file = fopen(argv[1], "rb");
+  RTC_CHECK(in_file) << "Could not open file " << argv[1] << " for reading";
+  printf("Timing file: %s\n", argv[2]);
+  std::ifstream timing_file(argv[2]);
+  printf("Output file: %s\n", argv[3]);
+  FILE* out_file = fopen(argv[3], "wb");
+  RTC_CHECK(out_file) << "Could not open file " << argv[2] << " for writing";
+
+  // Copy the RTP file header to the output file.
+  char header_string[30];
+  RTC_CHECK(fgets(header_string, 30, in_file));
+  fprintf(out_file, "%s", header_string);
+  uint8_t file_header[16];
+  RTC_CHECK_EQ(fread(file_header, sizeof(file_header), 1, in_file), 1);
+  RTC_CHECK_EQ(fwrite(file_header, sizeof(file_header), 1, out_file), 1);
+
+  // Read all time values from the timing file. Store in a vector.
+  std::vector<int> new_arrival_times;
+  int new_time;
+  while (timing_file >> new_time) {
+    new_arrival_times.push_back(new_time);
+  }
+
+  // Read all packets from the input RTP file, but no more than the number of
+  // new time values. Store RTP packets together with new time values.
+  auto time_it = new_arrival_times.begin();
+  std::vector<PacketAndTime> packets;
+  while (1) {
+    auto packet = ReadNextPacket(in_file);
+    if (packet.empty() || time_it == new_arrival_times.end()) {
+      break;
+    }
+    packets.push_back({std::move(packet), *time_it});
+    ++time_it;
+  }
+
+  // Sort on new time values.
+  std::sort(packets.begin(), packets.end(),
+            [](const PacketAndTime& a, const PacketAndTime& b) {
+              return a.time < b.time;
+            });
+
+  // Write packets to output file.
+  for (const auto& p : packets) {
+    WritePacket(p, out_file);
+  }
+
+  fclose(in_file);
+  fclose(out_file);
+  return 0;
+}
+
+}  // namespace
+}  // namespace test
+}  // namespace webrtc
+
+int main(int argc, char* argv[]) {
+  return webrtc::test::RunRtpJitter(argc, argv);
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtpcat.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtpcat.cc
new file mode 100644
index 0000000000..431de553ae
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/rtpcat.cc
@@ -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.
+ */
+
+#include <stdio.h>
+
+#include <memory>
+
+#include "rtc_base/checks.h"
+#include "test/rtp_file_reader.h"
+#include "test/rtp_file_writer.h"
+
+using webrtc::test::RtpFileReader;
+using webrtc::test::RtpFileWriter;
+
+int main(int argc, char* argv[]) {
+  if (argc < 3) {
+    printf("Concatenates multiple rtpdump files into one.\n");
+    printf("Usage: rtpcat in1.rtp int2.rtp [...] out.rtp\n");
+    exit(1);
+  }
+
+  std::unique_ptr<RtpFileWriter> output(
+      RtpFileWriter::Create(RtpFileWriter::kRtpDump, argv[argc - 1]));
+  RTC_CHECK(output.get() != NULL) << "Cannot open output file.";
+  printf("Output RTP file: %s\n", argv[argc - 1]);
+
+  for (int i = 1; i < argc - 1; i++) {
+    std::unique_ptr<RtpFileReader> input(
+        RtpFileReader::Create(RtpFileReader::kRtpDump, argv[i]));
+    RTC_CHECK(input.get() != NULL) << "Cannot open input file " << argv[i];
+    printf("Input RTP file: %s\n", argv[i]);
+
+    webrtc::test::RtpPacket packet;
+    while (input->NextPacket(&packet))
+      RTC_CHECK(output->WritePacket(&packet));
+  }
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.cc b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.cc
new file mode 100644
index 0000000000..baed812327
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/underrun_optimizer.h"
+
+#include <algorithm>
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDelayBuckets = 100;
+constexpr int kBucketSizeMs = 20;
+
+}  // namespace
+
+UnderrunOptimizer::UnderrunOptimizer(const TickTimer* tick_timer,
+                                     int histogram_quantile,
+                                     int forget_factor,
+                                     absl::optional<int> start_forget_weight,
+                                     absl::optional<int> resample_interval_ms)
+    : tick_timer_(tick_timer),
+      histogram_(kDelayBuckets, forget_factor, start_forget_weight),
+      histogram_quantile_(histogram_quantile),
+      resample_interval_ms_(resample_interval_ms) {}
+
+void UnderrunOptimizer::Update(int relative_delay_ms) {
+  absl::optional<int> histogram_update;
+  if (resample_interval_ms_) {
+    if (!resample_stopwatch_) {
+      resample_stopwatch_ = tick_timer_->GetNewStopwatch();
+    }
+    if (static_cast<int>(resample_stopwatch_->ElapsedMs()) >
+        *resample_interval_ms_) {
+      histogram_update = max_delay_in_interval_ms_;
+      resample_stopwatch_ = tick_timer_->GetNewStopwatch();
+      max_delay_in_interval_ms_ = 0;
+    }
+    max_delay_in_interval_ms_ =
+        std::max(max_delay_in_interval_ms_, relative_delay_ms);
+  } else {
+    histogram_update = relative_delay_ms;
+  }
+  if (!histogram_update) {
+    return;
+  }
+
+  const int index = *histogram_update / kBucketSizeMs;
+  if (index < histogram_.NumBuckets()) {
+    // Maximum delay to register is 2000 ms.
+    histogram_.Add(index);
+  }
+  int bucket_index = histogram_.Quantile(histogram_quantile_);
+  optimal_delay_ms_ = (1 + bucket_index) * kBucketSizeMs;
+}
+
+void UnderrunOptimizer::Reset() {
+  histogram_.Reset();
+  resample_stopwatch_.reset();
+  max_delay_in_interval_ms_ = 0;
+  optimal_delay_ms_.reset();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h
new file mode 100644
index 0000000000..b37ce18795
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
+#define MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/histogram.h"
+
+namespace webrtc {
+
+// Estimates probability of buffer underrun due to late packet arrival.
+// The optimal delay is decided such that the probability of underrun is lower
+// than 1 - `histogram_quantile`.
+class UnderrunOptimizer {
+ public:
+  UnderrunOptimizer(const TickTimer* tick_timer,
+                    int histogram_quantile,
+                    int forget_factor,
+                    absl::optional<int> start_forget_weight,
+                    absl::optional<int> resample_interval_ms);
+
+  void Update(int relative_delay_ms);
+
+  absl::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }
+
+  void Reset();
+
+ private:
+  const TickTimer* tick_timer_;
+  Histogram histogram_;
+  const int histogram_quantile_;  // In Q30.
+  const absl::optional<int> resample_interval_ms_;
+  std::unique_ptr<TickTimer::Stopwatch> resample_stopwatch_;
+  int max_delay_in_interval_ms_ = 0;
+  absl::optional<int> optimal_delay_ms_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer_unittest.cc
new file mode 100644
index 0000000000..a86e9cf107
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer_unittest.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/underrun_optimizer.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kDefaultHistogramQuantile = 1020054733;  // 0.95 in Q30.
+constexpr int kForgetFactor = 32745;                   // 0.9993 in Q15.
+
+}  // namespace
+
+TEST(UnderrunOptimizerTest, ResamplePacketDelays) {
+  TickTimer tick_timer;
+  constexpr int kResampleIntervalMs = 500;
+  UnderrunOptimizer underrun_optimizer(&tick_timer, kDefaultHistogramQuantile,
+                                       kForgetFactor, absl::nullopt,
+                                       kResampleIntervalMs);
+
+  // The histogram should be updated once with the maximum delay observed for
+  // the following sequence of updates.
+  for (int i = 0; i < 500; i += 20) {
+    underrun_optimizer.Update(i);
+    EXPECT_FALSE(underrun_optimizer.GetOptimalDelayMs());
+  }
+  tick_timer.Increment(kResampleIntervalMs / tick_timer.ms_per_tick() + 1);
+  underrun_optimizer.Update(0);
+  EXPECT_EQ(underrun_optimizer.GetOptimalDelayMs(), 500);
+}
+
+}  // namespace webrtc