Adding upstream version 124.0.1.upstream/124.0.1

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

32 files changed, 7791 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/DEPS b/third_party/libwebrtc/modules/audio_coding/codecs/opus/DEPS
new file mode 100644
index 0000000000..c2530726ad
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/DEPS
@@ -0,0 +1,5 @@
+specific_include_rules = {
+  "opus_inst\.h": [
+    "+third_party/opus",
+  ],
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc
new file mode 100644
index 0000000000..03c02186d0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.cc
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
+                                               absl::string_view param) {
+  auto it = format.parameters.find(std::string(param));
+  if (it == format.parameters.end())
+    return absl::nullopt;
+
+  return it->second;
+}
+
+// Parses a comma-separated string "1,2,0,6" into a std::vector<unsigned char>.
+template <>
+absl::optional<std::vector<unsigned char>> GetFormatParameter(
+    const SdpAudioFormat& format,
+    absl::string_view param) {
+  std::vector<unsigned char> result;
+  const std::string comma_separated_list =
+      GetFormatParameter(format, param).value_or("");
+  size_t pos = 0;
+  while (pos < comma_separated_list.size()) {
+    const size_t next_comma = comma_separated_list.find(',', pos);
+    const size_t distance_to_next_comma = next_comma == std::string::npos
+                                              ? std::string::npos
+                                              : (next_comma - pos);
+    auto substring_with_number =
+        comma_separated_list.substr(pos, distance_to_next_comma);
+    auto conv = rtc::StringToNumber<int>(substring_with_number);
+    if (!conv.has_value()) {
+      return absl::nullopt;
+    }
+    result.push_back(*conv);
+    pos += substring_with_number.size() + 1;
+  }
+  return result;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
new file mode 100644
index 0000000000..5ebb51b577
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_coder_opus_common.h
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
+
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
+                                               absl::string_view param);
+
+template <typename T>
+absl::optional<T> GetFormatParameter(const SdpAudioFormat& format,
+                                     absl::string_view param) {
+  return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or(""));
+}
+
+template <>
+absl::optional<std::vector<unsigned char>> GetFormatParameter(
+    const SdpAudioFormat& format,
+    absl::string_view param);
+
+class OpusFrame : public AudioDecoder::EncodedAudioFrame {
+ public:
+  OpusFrame(AudioDecoder* decoder,
+            rtc::Buffer&& payload,
+            bool is_primary_payload)
+      : decoder_(decoder),
+        payload_(std::move(payload)),
+        is_primary_payload_(is_primary_payload) {}
+
+  size_t Duration() const override {
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->PacketDuration(payload_.data(), payload_.size());
+    } else {
+      ret = decoder_->PacketDurationRedundant(payload_.data(), payload_.size());
+    }
+    return (ret < 0) ? 0 : static_cast<size_t>(ret);
+  }
+
+  bool IsDtxPacket() const override { return payload_.size() <= 2; }
+
+  absl::optional<DecodeResult> Decode(
+      rtc::ArrayView<int16_t> decoded) const override {
+    AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+    int ret;
+    if (is_primary_payload_) {
+      ret = decoder_->Decode(
+          payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+          decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    } else {
+      ret = decoder_->DecodeRedundant(
+          payload_.data(), payload_.size(), decoder_->SampleRateHz(),
+          decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
+    }
+
+    if (ret < 0)
+      return absl::nullopt;
+
+    return DecodeResult{static_cast<size_t>(ret), speech_type};
+  }
+
+ private:
+  AudioDecoder* const decoder_;
+  const rtc::Buffer payload_;
+  const bool is_primary_payload_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_CODER_OPUS_COMMON_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
new file mode 100644
index 0000000000..285ea89959
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.cc
@@ -0,0 +1,182 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h"
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+std::unique_ptr<AudioDecoderMultiChannelOpusImpl>
+AudioDecoderMultiChannelOpusImpl::MakeAudioDecoder(
+    AudioDecoderMultiChannelOpusConfig config) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  // Fill the pointer with a working decoder through the C interface. This
+  // allocates memory.
+  OpusDecInst* dec_state = nullptr;
+  const int error = WebRtcOpus_MultistreamDecoderCreate(
+      &dec_state, config.num_channels, config.num_streams,
+      config.coupled_streams, config.channel_mapping.data());
+  if (error != 0) {
+    return nullptr;
+  }
+
+  // Pass the ownership to DecoderImpl. Not using 'make_unique' because the
+  // c-tor is private.
+  return std::unique_ptr<AudioDecoderMultiChannelOpusImpl>(
+      new AudioDecoderMultiChannelOpusImpl(dec_state, config));
+}
+
+AudioDecoderMultiChannelOpusImpl::AudioDecoderMultiChannelOpusImpl(
+    OpusDecInst* dec_state,
+    AudioDecoderMultiChannelOpusConfig config)
+    : dec_state_(dec_state), config_(config) {
+  RTC_DCHECK(dec_state);
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderMultiChannelOpusImpl::~AudioDecoderMultiChannelOpusImpl() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+absl::optional<AudioDecoderMultiChannelOpusConfig>
+AudioDecoderMultiChannelOpusImpl::SdpToConfig(const SdpAudioFormat& format) {
+  AudioDecoderMultiChannelOpusConfig config;
+  config.num_channels = format.num_channels;
+  auto num_streams = GetFormatParameter<int>(format, "num_streams");
+  if (!num_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.num_streams = *num_streams;
+
+  auto coupled_streams = GetFormatParameter<int>(format, "coupled_streams");
+  if (!coupled_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.coupled_streams = *coupled_streams;
+
+  auto channel_mapping =
+      GetFormatParameter<std::vector<unsigned char>>(format, "channel_mapping");
+  if (!channel_mapping.has_value()) {
+    return absl::nullopt;
+  }
+  config.channel_mapping = *channel_mapping;
+  if (!config.IsOk()) {
+    return absl::nullopt;
+  }
+  return config;
+}
+
+std::vector<AudioDecoder::ParseResult>
+AudioDecoderMultiChannelOpusImpl::ParsePayload(rtc::Buffer&& payload,
+                                               uint32_t timestamp) {
+  std::vector<ParseResult> results;
+
+  if (PacketHasFec(payload.data(), payload.size())) {
+    const int duration =
+        PacketDurationRedundant(payload.data(), payload.size());
+    RTC_DCHECK_GE(duration, 0);
+    rtc::Buffer payload_copy(payload.data(), payload.size());
+    std::unique_ptr<EncodedAudioFrame> fec_frame(
+        new OpusFrame(this, std::move(payload_copy), false));
+    results.emplace_back(timestamp - duration, 1, std::move(fec_frame));
+  }
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OpusFrame(this, std::move(payload), true));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int AudioDecoderMultiChannelOpusImpl::DecodeInternal(const uint8_t* encoded,
+                                                     size_t encoded_len,
+                                                     int sample_rate_hz,
+                                                     int16_t* decoded,
+                                                     SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(
+        config_.num_channels);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderMultiChannelOpusImpl::DecodeRedundantInternal(
+    const uint8_t* encoded,
+    size_t encoded_len,
+    int sample_rate_hz,
+    int16_t* decoded,
+    SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  RTC_DCHECK_EQ(sample_rate_hz, 48000);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(
+        config_.num_channels);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderMultiChannelOpusImpl::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderMultiChannelOpusImpl::PacketDuration(const uint8_t* encoded,
+                                                     size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderMultiChannelOpusImpl::PacketDurationRedundant(
+    const uint8_t* encoded,
+    size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len, 48000);
+}
+
+bool AudioDecoderMultiChannelOpusImpl::PacketHasFec(const uint8_t* encoded,
+                                                    size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+int AudioDecoderMultiChannelOpusImpl::SampleRateHz() const {
+  return 48000;
+}
+
+size_t AudioDecoderMultiChannelOpusImpl::Channels() const {
+  return config_.num_channels;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
new file mode 100644
index 0000000000..2ff47a8a53
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoderMultiChannelOpusImpl final : public AudioDecoder {
+ public:
+  static std::unique_ptr<AudioDecoderMultiChannelOpusImpl> MakeAudioDecoder(
+      AudioDecoderMultiChannelOpusConfig config);
+
+  ~AudioDecoderMultiChannelOpusImpl() override;
+
+  AudioDecoderMultiChannelOpusImpl(const AudioDecoderMultiChannelOpusImpl&) =
+      delete;
+  AudioDecoderMultiChannelOpusImpl& operator=(
+      const AudioDecoderMultiChannelOpusImpl&) = delete;
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+  static absl::optional<AudioDecoderMultiChannelOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  int DecodeRedundantInternal(const uint8_t* encoded,
+                              size_t encoded_len,
+                              int sample_rate_hz,
+                              int16_t* decoded,
+                              SpeechType* speech_type) override;
+
+ private:
+  AudioDecoderMultiChannelOpusImpl(OpusDecInst* dec_state,
+                                   AudioDecoderMultiChannelOpusConfig config);
+
+  OpusDecInst* dec_state_;
+  const AudioDecoderMultiChannelOpusConfig config_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_MULTI_CHANNEL_OPUS_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc
new file mode 100644
index 0000000000..57e2107f3c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_unittest.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+using ::testing::NiceMock;
+using ::testing::Return;
+
+TEST(AudioDecoderMultiOpusTest, GetFormatParameter) {
+  const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                  {{"channel_mapping", "0,1,2,3"},
+                                   {"coupled_streams", "2"},
+                                   {"num_streams", "2"}});
+
+  EXPECT_EQ(GetFormatParameter(sdp_format, "channel_mapping"),
+            absl::optional<std::string>("0,1,2,3"));
+
+  EXPECT_EQ(GetFormatParameter<int>(sdp_format, "coupled_streams"),
+            absl::optional<int>(2));
+
+  EXPECT_EQ(GetFormatParameter(sdp_format, "missing"), absl::nullopt);
+
+  EXPECT_EQ(GetFormatParameter<int>(sdp_format, "channel_mapping"),
+            absl::nullopt);
+}
+
+TEST(AudioDecoderMultiOpusTest, InvalidChannelMappings) {
+  {
+    // Can't use channel 3 if there are only 2 channels.
+    const SdpAudioFormat sdp_format("multiopus", 48000, 2,
+                                    {{"channel_mapping", "3,0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+  {
+    // The mapping is too long. There are only 5 channels, but 6 elements in the
+    // mapping.
+    const SdpAudioFormat sdp_format("multiopus", 48000, 5,
+                                    {{"channel_mapping", "0,1,2,3,4,5"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+  {
+    // The mapping doesn't parse correctly.
+    const SdpAudioFormat sdp_format(
+        "multiopus", 48000, 5,
+        {{"channel_mapping", "0,1,two,3,4"}, {"coupled_streams", "0"}});
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+}
+
+TEST(AudioDecoderMultiOpusTest, ValidSdpToConfigProducesCorrectConfig) {
+  const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                  {{"channel_mapping", "3,1,2,0"},
+                                   {"coupled_streams", "2"},
+                                   {"num_streams", "2"}});
+
+  const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+      AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+  ASSERT_TRUE(decoder_config.has_value());
+  EXPECT_TRUE(decoder_config->IsOk());
+  EXPECT_EQ(decoder_config->coupled_streams, 2);
+  EXPECT_THAT(decoder_config->channel_mapping,
+              ::testing::ContainerEq(std::vector<unsigned char>({3, 1, 2, 0})));
+}
+
+TEST(AudioDecoderMultiOpusTest, InvalidSdpToConfigDoesNotProduceConfig) {
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                    {{"channel_mapping", "0,1,2,3"},
+                                     {"coupled_stream", "2"},
+                                     {"num_streams", "2"}});
+
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                    {{"channel_mapping", "0,1,2 3"},
+                                     {"coupled_streams", "2"},
+                                     {"num_streams", "2"}});
+
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    EXPECT_FALSE(decoder_config.has_value());
+  }
+}
+
+TEST(AudioDecoderMultiOpusTest, CodecsCanBeCreated) {
+  const SdpAudioFormat sdp_format("multiopus", 48000, 4,
+                                  {{"channel_mapping", "0,1,2,3"},
+                                   {"coupled_streams", "2"},
+                                   {"num_streams", "2"}});
+
+  const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+      AudioDecoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+  ASSERT_TRUE(decoder_config.has_value());
+
+  const std::unique_ptr<AudioDecoder> opus_decoder =
+      AudioDecoderMultiChannelOpus::MakeAudioDecoder(*decoder_config);
+
+  EXPECT_TRUE(opus_decoder);
+}
+
+TEST(AudioDecoderMultiOpusTest, AdvertisedCodecsCanBeCreated) {
+  std::vector<AudioCodecSpec> specs;
+  AudioDecoderMultiChannelOpus::AppendSupportedDecoders(&specs);
+
+  EXPECT_FALSE(specs.empty());
+
+  for (const AudioCodecSpec& spec : specs) {
+    const absl::optional<AudioDecoderMultiChannelOpus::Config> decoder_config =
+        AudioDecoderMultiChannelOpus::SdpToConfig(spec.format);
+    ASSERT_TRUE(decoder_config.has_value());
+
+    const std::unique_ptr<AudioDecoder> opus_decoder =
+        AudioDecoderMultiChannelOpus::MakeAudioDecoder(*decoder_config);
+
+    EXPECT_TRUE(opus_decoder);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
new file mode 100644
index 0000000000..cff9685548
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels,
+                                           int sample_rate_hz)
+    : channels_{num_channels}, sample_rate_hz_{sample_rate_hz} {
+  RTC_DCHECK(num_channels == 1 || num_channels == 2);
+  RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000);
+  const int error =
+      WebRtcOpus_DecoderCreate(&dec_state_, channels_, sample_rate_hz_);
+  RTC_DCHECK(error == 0);
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+AudioDecoderOpusImpl::~AudioDecoderOpusImpl() {
+  WebRtcOpus_DecoderFree(dec_state_);
+}
+
+std::vector<AudioDecoder::ParseResult> AudioDecoderOpusImpl::ParsePayload(
+    rtc::Buffer&& payload,
+    uint32_t timestamp) {
+  std::vector<ParseResult> results;
+
+  if (PacketHasFec(payload.data(), payload.size())) {
+    const int duration =
+        PacketDurationRedundant(payload.data(), payload.size());
+    RTC_DCHECK_GE(duration, 0);
+    rtc::Buffer payload_copy(payload.data(), payload.size());
+    std::unique_ptr<EncodedAudioFrame> fec_frame(
+        new OpusFrame(this, std::move(payload_copy), false));
+    results.emplace_back(timestamp - duration, 1, std::move(fec_frame));
+  }
+  std::unique_ptr<EncodedAudioFrame> frame(
+      new OpusFrame(this, std::move(payload), true));
+  results.emplace_back(timestamp, 0, std::move(frame));
+  return results;
+}
+
+int AudioDecoderOpusImpl::DecodeInternal(const uint8_t* encoded,
+                                         size_t encoded_len,
+                                         int sample_rate_hz,
+                                         int16_t* decoded,
+                                         SpeechType* speech_type) {
+  RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret =
+      WebRtcOpus_Decode(dec_state_, encoded, encoded_len, decoded, &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+int AudioDecoderOpusImpl::DecodeRedundantInternal(const uint8_t* encoded,
+                                                  size_t encoded_len,
+                                                  int sample_rate_hz,
+                                                  int16_t* decoded,
+                                                  SpeechType* speech_type) {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return DecodeInternal(encoded, encoded_len, sample_rate_hz, decoded,
+                          speech_type);
+  }
+
+  RTC_DCHECK_EQ(sample_rate_hz, sample_rate_hz_);
+  int16_t temp_type = 1;  // Default is speech.
+  int ret = WebRtcOpus_DecodeFec(dec_state_, encoded, encoded_len, decoded,
+                                 &temp_type);
+  if (ret > 0)
+    ret *= static_cast<int>(channels_);  // Return total number of samples.
+  *speech_type = ConvertSpeechType(temp_type);
+  return ret;
+}
+
+void AudioDecoderOpusImpl::Reset() {
+  WebRtcOpus_DecoderInit(dec_state_);
+}
+
+int AudioDecoderOpusImpl::PacketDuration(const uint8_t* encoded,
+                                         size_t encoded_len) const {
+  return WebRtcOpus_DurationEst(dec_state_, encoded, encoded_len);
+}
+
+int AudioDecoderOpusImpl::PacketDurationRedundant(const uint8_t* encoded,
+                                                  size_t encoded_len) const {
+  if (!PacketHasFec(encoded, encoded_len)) {
+    // This packet is a RED packet.
+    return PacketDuration(encoded, encoded_len);
+  }
+
+  return WebRtcOpus_FecDurationEst(encoded, encoded_len, sample_rate_hz_);
+}
+
+bool AudioDecoderOpusImpl::PacketHasFec(const uint8_t* encoded,
+                                        size_t encoded_len) const {
+  int fec;
+  fec = WebRtcOpus_PacketHasFec(encoded, encoded_len);
+  return (fec == 1);
+}
+
+int AudioDecoderOpusImpl::SampleRateHz() const {
+  return sample_rate_hz_;
+}
+
+size_t AudioDecoderOpusImpl::Channels() const {
+  return channels_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
new file mode 100644
index 0000000000..e8fd0440bc
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+
+class AudioDecoderOpusImpl final : public AudioDecoder {
+ public:
+  explicit AudioDecoderOpusImpl(size_t num_channels,
+                                int sample_rate_hz = 48000);
+  ~AudioDecoderOpusImpl() override;
+
+  AudioDecoderOpusImpl(const AudioDecoderOpusImpl&) = delete;
+  AudioDecoderOpusImpl& operator=(const AudioDecoderOpusImpl&) = delete;
+
+  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                        uint32_t timestamp) override;
+  void Reset() override;
+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
+  int SampleRateHz() const override;
+  size_t Channels() const override;
+
+ protected:
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+  int DecodeRedundantInternal(const uint8_t* encoded,
+                              size_t encoded_len,
+                              int sample_rate_hz,
+                              int16_t* decoded,
+                              SpeechType* speech_type) override;
+
+ private:
+  OpusDecInst* dec_state_;
+  const size_t channels_;
+  const int sample_rate_hz_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
new file mode 100644
index 0000000000..38a11c123d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
@@ -0,0 +1,366 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+/*
+ * LEFT TO DO:
+ * - WRITE TESTS for the stuff in this  file.
+ * - Check the creation, maybe make it safer by returning an empty optional or
+ *   unique_ptr. --- It looks OK, but RecreateEncoderInstance can perhaps crash
+ *   on a valid config. Can run it in the fuzzer for some time. Should prbl also
+ *   fuzz the config.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h"
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+namespace {
+
+// Recommended bitrates for one channel:
+// 8-12 kb/s for NB speech,
+// 16-20 kb/s for WB speech,
+// 28-40 kb/s for FB speech,
+// 48-64 kb/s for FB mono music, and
+// 64-128 kb/s for FB stereo music.
+// The current implementation multiplies these values by the number of channels.
+constexpr int kOpusBitrateNbBps = 12000;
+constexpr int kOpusBitrateWbBps = 20000;
+constexpr int kOpusBitrateFbBps = 32000;
+
+constexpr int kDefaultMaxPlaybackRate = 48000;
+// These two lists must be sorted from low to high
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60, 120};
+#else
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60};
+#endif
+
+int GetBitrateBps(const AudioEncoderMultiChannelOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return config.bitrate_bps;
+}
+int GetMaxPlaybackRate(const SdpAudioFormat& format) {
+  const auto param = GetFormatParameter<int>(format, "maxplaybackrate");
+  if (param && *param >= 8000) {
+    return std::min(*param, kDefaultMaxPlaybackRate);
+  }
+  return kDefaultMaxPlaybackRate;
+}
+
+int GetFrameSizeMs(const SdpAudioFormat& format) {
+  const auto ptime = GetFormatParameter<int>(format, "ptime");
+  if (ptime.has_value()) {
+    // Pick the next highest supported frame length from
+    // kOpusSupportedFrameLengths.
+    for (const int supported_frame_length : kOpusSupportedFrameLengths) {
+      if (supported_frame_length >= *ptime) {
+        return supported_frame_length;
+      }
+    }
+    // If none was found, return the largest supported frame length.
+    return *(std::end(kOpusSupportedFrameLengths) - 1);
+  }
+
+  return AudioEncoderOpusConfig::kDefaultFrameSizeMs;
+}
+
+int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
+  const int bitrate = [&] {
+    if (max_playback_rate <= 8000) {
+      return kOpusBitrateNbBps * rtc::dchecked_cast<int>(num_channels);
+    } else if (max_playback_rate <= 16000) {
+      return kOpusBitrateWbBps * rtc::dchecked_cast<int>(num_channels);
+    } else {
+      return kOpusBitrateFbBps * rtc::dchecked_cast<int>(num_channels);
+    }
+  }();
+  RTC_DCHECK_GE(bitrate, AudioEncoderMultiChannelOpusConfig::kMinBitrateBps);
+  return bitrate;
+}
+
+// Get the maxaveragebitrate parameter in string-form, so we can properly figure
+// out how invalid it is and accurately log invalid values.
+int CalculateBitrate(int max_playback_rate_hz,
+                     size_t num_channels,
+                     absl::optional<std::string> bitrate_param) {
+  const int default_bitrate =
+      CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
+
+  if (bitrate_param) {
+    const auto bitrate = rtc::StringToNumber<int>(*bitrate_param);
+    if (bitrate) {
+      const int chosen_bitrate =
+          std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                   std::min(*bitrate, AudioEncoderOpusConfig::kMaxBitrateBps));
+      if (bitrate != chosen_bitrate) {
+        RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate
+                            << " clamped to " << chosen_bitrate;
+      }
+      return chosen_bitrate;
+    }
+    RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate \"" << *bitrate_param
+                        << "\" replaced by default bitrate " << default_bitrate;
+  }
+
+  return default_bitrate;
+}
+
+}  // namespace
+
+std::unique_ptr<AudioEncoder>
+AudioEncoderMultiChannelOpusImpl::MakeAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config,
+    int payload_type) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderMultiChannelOpusImpl>(config,
+                                                            payload_type);
+}
+
+AudioEncoderMultiChannelOpusImpl::AudioEncoderMultiChannelOpusImpl(
+    const AudioEncoderMultiChannelOpusConfig& config,
+    int payload_type)
+    : payload_type_(payload_type), inst_(nullptr) {
+  RTC_DCHECK(0 <= payload_type && payload_type <= 127);
+
+  RTC_CHECK(RecreateEncoderInstance(config));
+}
+
+AudioEncoderMultiChannelOpusImpl::~AudioEncoderMultiChannelOpusImpl() {
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+}
+
+size_t AudioEncoderMultiChannelOpusImpl::SufficientOutputBufferSize() const {
+  // Calculate the number of bytes we expect the encoder to produce,
+  // then multiply by two to give a wide margin for error.
+  const size_t bytes_per_millisecond =
+      static_cast<size_t>(GetBitrateBps(config_) / (1000 * 8) + 1);
+  const size_t approx_encoded_bytes =
+      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
+  return 2 * approx_encoded_bytes;
+}
+
+void AudioEncoderMultiChannelOpusImpl::Reset() {
+  RTC_CHECK(RecreateEncoderInstance(config_));
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta>>
+AudioEncoderMultiChannelOpusImpl::GetFrameLengthRange() const {
+  return {{TimeDelta::Millis(config_.frame_size_ms),
+           TimeDelta::Millis(config_.frame_size_ms)}};
+}
+
+// If the given config is OK, recreate the Opus encoder instance with those
+// settings, save the config, and return true. Otherwise, do nothing and return
+// false.
+bool AudioEncoderMultiChannelOpusImpl::RecreateEncoderInstance(
+    const AudioEncoderMultiChannelOpusConfig& config) {
+  if (!config.IsOk())
+    return false;
+  config_ = config;
+  if (inst_)
+    RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+  input_buffer_.clear();
+  input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_MultistreamEncoderCreate(
+             &inst_, config.num_channels,
+             config.application ==
+                     AudioEncoderMultiChannelOpusConfig::ApplicationMode::kVoip
+                 ? 0
+                 : 1,
+             config.num_streams, config.coupled_streams,
+             config.channel_mapping.data()));
+  const int bitrate = GetBitrateBps(config);
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, bitrate));
+  RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
+  if (config.fec_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus enable FEC";
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus disable FEC";
+  }
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
+  RTC_LOG(LS_VERBOSE) << "Set Opus playback rate to "
+                      << config.max_playback_rate_hz << " hz.";
+
+  // Use the DEFAULT complexity.
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_SetComplexity(inst_, AudioEncoderOpusConfig().complexity));
+  RTC_LOG(LS_VERBOSE) << "Set Opus coding complexity to "
+                      << AudioEncoderOpusConfig().complexity;
+
+  if (config.dtx_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus enable DTX";
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus disable DTX";
+  }
+
+  if (config.cbr_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableCbr(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus enable CBR";
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableCbr(inst_));
+    RTC_LOG(LS_VERBOSE) << "Opus disable CBR";
+  }
+  num_channels_to_encode_ = NumChannels();
+  next_frame_length_ms_ = config_.frame_size_ms;
+  RTC_LOG(LS_VERBOSE) << "Set Opus frame length to " << config_.frame_size_ms
+                      << " ms";
+  return true;
+}
+
+absl::optional<AudioEncoderMultiChannelOpusConfig>
+AudioEncoderMultiChannelOpusImpl::SdpToConfig(const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name, "multiopus") ||
+      format.clockrate_hz != 48000) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderMultiChannelOpusConfig config;
+  config.num_channels = format.num_channels;
+  config.frame_size_ms = GetFrameSizeMs(format);
+  config.max_playback_rate_hz = GetMaxPlaybackRate(format);
+  config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
+  config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
+  config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
+  config.bitrate_bps =
+      CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
+                       GetFormatParameter(format, "maxaveragebitrate"));
+  config.application =
+      config.num_channels == 1
+          ? AudioEncoderMultiChannelOpusConfig::ApplicationMode::kVoip
+          : AudioEncoderMultiChannelOpusConfig::ApplicationMode::kAudio;
+
+  config.supported_frame_lengths_ms.clear();
+  std::copy(std::begin(kOpusSupportedFrameLengths),
+            std::end(kOpusSupportedFrameLengths),
+            std::back_inserter(config.supported_frame_lengths_ms));
+
+  auto num_streams = GetFormatParameter<int>(format, "num_streams");
+  if (!num_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.num_streams = *num_streams;
+
+  auto coupled_streams = GetFormatParameter<int>(format, "coupled_streams");
+  if (!coupled_streams.has_value()) {
+    return absl::nullopt;
+  }
+  config.coupled_streams = *coupled_streams;
+
+  auto channel_mapping =
+      GetFormatParameter<std::vector<unsigned char>>(format, "channel_mapping");
+  if (!channel_mapping.has_value()) {
+    return absl::nullopt;
+  }
+  config.channel_mapping = *channel_mapping;
+
+  if (!config.IsOk()) {
+    return absl::nullopt;
+  }
+  return config;
+}
+
+AudioCodecInfo AudioEncoderMultiChannelOpusImpl::QueryAudioEncoder(
+    const AudioEncoderMultiChannelOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  AudioCodecInfo info(48000, config.num_channels, config.bitrate_bps,
+                      AudioEncoderOpusConfig::kMinBitrateBps,
+                      AudioEncoderOpusConfig::kMaxBitrateBps);
+  info.allow_comfort_noise = false;
+  info.supports_network_adaption = false;
+  return info;
+}
+
+size_t AudioEncoderMultiChannelOpusImpl::Num10msFramesPerPacket() const {
+  return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
+}
+size_t AudioEncoderMultiChannelOpusImpl::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(48000, 100) * config_.num_channels;
+}
+int AudioEncoderMultiChannelOpusImpl::SampleRateHz() const {
+  return 48000;
+}
+size_t AudioEncoderMultiChannelOpusImpl::NumChannels() const {
+  return config_.num_channels;
+}
+size_t AudioEncoderMultiChannelOpusImpl::Num10MsFramesInNextPacket() const {
+  return Num10msFramesPerPacket();
+}
+size_t AudioEncoderMultiChannelOpusImpl::Max10MsFramesInAPacket() const {
+  return Num10msFramesPerPacket();
+}
+int AudioEncoderMultiChannelOpusImpl::GetTargetBitrate() const {
+  return GetBitrateBps(config_);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderMultiChannelOpusImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  if (input_buffer_.empty())
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
+  if (input_buffer_.size() <
+      (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(input_buffer_.size(),
+               Num10msFramesPerPacket() * SamplesPer10msFrame());
+
+  const size_t max_encoded_bytes = SufficientOutputBufferSize();
+  EncodedInfo info;
+  info.encoded_bytes = encoded->AppendData(
+      max_encoded_bytes, [&](rtc::ArrayView<uint8_t> encoded) {
+        int status = WebRtcOpus_Encode(
+            inst_, &input_buffer_[0],
+            rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
+            rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded.data());
+
+        RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
+
+        return static_cast<size_t>(status);
+      });
+  input_buffer_.clear();
+
+  // Will use new packet size for next encoding.
+  config_.frame_size_ms = next_frame_length_ms_;
+
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.send_even_if_empty = true;  // Allows Opus to send empty packets.
+
+  info.speech = true;
+  info.encoder_type = CodecType::kOther;
+
+  return info;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
new file mode 100644
index 0000000000..8a7210515c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
+#include "api/units/time_delta.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class AudioEncoderMultiChannelOpusImpl final : public AudioEncoder {
+ public:
+  AudioEncoderMultiChannelOpusImpl(
+      const AudioEncoderMultiChannelOpusConfig& config,
+      int payload_type);
+  ~AudioEncoderMultiChannelOpusImpl() override;
+
+  AudioEncoderMultiChannelOpusImpl(const AudioEncoderMultiChannelOpusImpl&) =
+      delete;
+  AudioEncoderMultiChannelOpusImpl& operator=(
+      const AudioEncoderMultiChannelOpusImpl&) = delete;
+
+  // Static interface for use by BuiltinAudioEncoderFactory.
+  static constexpr const char* GetPayloadName() { return "multiopus"; }
+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
+      const SdpAudioFormat& format);
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  void Reset() override;
+  absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
+      const override;
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  static absl::optional<AudioEncoderMultiChannelOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+  static AudioCodecInfo QueryAudioEncoder(
+      const AudioEncoderMultiChannelOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderMultiChannelOpusConfig&,
+      int payload_type);
+
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
+  size_t SufficientOutputBufferSize() const;
+  bool RecreateEncoderInstance(
+      const AudioEncoderMultiChannelOpusConfig& config);
+  void SetFrameLength(int frame_length_ms);
+  void SetNumChannelsToEncode(size_t num_channels_to_encode);
+  void SetProjectedPacketLossRate(float fraction);
+
+  AudioEncoderMultiChannelOpusConfig config_;
+  const int payload_type_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+  size_t num_channels_to_encode_;
+  int next_frame_length_ms_;
+
+  friend struct AudioEncoderMultiChannelOpus;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc
new file mode 100644
index 0000000000..92f6f2c169
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_unittest.cc
@@ -0,0 +1,156 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_multi_channel_opus.h"
+
+#include "test/gmock.h"
+
+namespace webrtc {
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace {
+constexpr int kOpusPayloadType = 120;
+}  // namespace
+
+TEST(AudioEncoderMultiOpusTest, CheckConfigValidity) {
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 2,
+                                    {{"channel_mapping", "3,0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // Maps input channel 0 to coded channel 3, which doesn't exist.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 2,
+                                    {{"channel_mapping", "0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // The mapping is too short.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,0,0"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "1"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // Coded channel 0 comes from both input channels 0, 1 and 2.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,255,255"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "1"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+    ASSERT_TRUE(encoder_config.has_value());
+
+    // This is fine, because channels 1, 2 are set to be ignored.
+    EXPECT_TRUE(encoder_config->IsOk());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,255,255"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+
+    // This is NOT fine, because channels nothing says how coded channel 1
+    // should be coded.
+    EXPECT_FALSE(encoder_config.has_value());
+  }
+}
+
+TEST(AudioEncoderMultiOpusTest, ConfigValuesAreParsedCorrectly) {
+  SdpAudioFormat sdp_format({"multiopus",
+                             48000,
+                             6,
+                             {{"minptime", "10"},
+                              {"useinbandfec", "1"},
+                              {"channel_mapping", "0,4,1,2,3,5"},
+                              {"num_streams", "4"},
+                              {"coupled_streams", "2"}}});
+  const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+      AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+  ASSERT_TRUE(encoder_config.has_value());
+
+  EXPECT_EQ(encoder_config->coupled_streams, 2);
+  EXPECT_EQ(encoder_config->num_streams, 4);
+  EXPECT_THAT(
+      encoder_config->channel_mapping,
+      testing::ContainerEq(std::vector<unsigned char>({0, 4, 1, 2, 3, 5})));
+}
+
+TEST(AudioEncoderMultiOpusTest, CreateFromValidConfig) {
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "0,255,255"},
+                                     {"coupled_streams", "0"},
+                                     {"num_streams", "2"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+    ASSERT_FALSE(encoder_config.has_value());
+  }
+  {
+    const SdpAudioFormat sdp_format("multiopus", 48000, 3,
+                                    {{"channel_mapping", "1,255,0"},
+                                     {"coupled_streams", "1"},
+                                     {"num_streams", "1"}});
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(sdp_format);
+    ASSERT_TRUE(encoder_config.has_value());
+
+    EXPECT_THAT(encoder_config->channel_mapping,
+                testing::ContainerEq(std::vector<unsigned char>({1, 255, 0})));
+
+    EXPECT_TRUE(encoder_config->IsOk());
+
+    const std::unique_ptr<AudioEncoder> opus_encoder =
+        AudioEncoderMultiChannelOpus::MakeAudioEncoder(*encoder_config,
+                                                       kOpusPayloadType);
+
+    // Creating an encoder from a valid config should work.
+    EXPECT_TRUE(opus_encoder);
+  }
+}
+
+TEST(AudioEncoderMultiOpusTest, AdvertisedCodecsCanBeCreated) {
+  std::vector<AudioCodecSpec> specs;
+  AudioEncoderMultiChannelOpus::AppendSupportedEncoders(&specs);
+
+  EXPECT_FALSE(specs.empty());
+
+  for (const AudioCodecSpec& spec : specs) {
+    const absl::optional<AudioEncoderMultiChannelOpus::Config> encoder_config =
+        AudioEncoderMultiChannelOpus::SdpToConfig(spec.format);
+    ASSERT_TRUE(encoder_config.has_value());
+
+    const std::unique_ptr<AudioEncoder> opus_encoder =
+        AudioEncoderMultiChannelOpus::MakeAudioEncoder(*encoder_config,
+                                                       kOpusPayloadType);
+
+    EXPECT_TRUE(opus_encoder);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
new file mode 100644
index 0000000000..17e0e33b1d
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -0,0 +1,824 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+
+#include <algorithm>
+#include <iterator>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/exp_filter.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/string_to_number.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+
+// Codec parameters for Opus.
+// draft-spittka-payload-rtp-opus-03
+
+// Recommended bitrates:
+// 8-12 kb/s for NB speech,
+// 16-20 kb/s for WB speech,
+// 28-40 kb/s for FB speech,
+// 48-64 kb/s for FB mono music, and
+// 64-128 kb/s for FB stereo music.
+// The current implementation applies the following values to mono signals,
+// and multiplies them by 2 for stereo.
+constexpr int kOpusBitrateNbBps = 12000;
+constexpr int kOpusBitrateWbBps = 20000;
+constexpr int kOpusBitrateFbBps = 32000;
+
+constexpr int kRtpTimestampRateHz = 48000;
+constexpr int kDefaultMaxPlaybackRate = 48000;
+
+// These two lists must be sorted from low to high
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+constexpr int kANASupportedFrameLengths[] = {20, 40, 60, 120};
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60, 120};
+#else
+constexpr int kANASupportedFrameLengths[] = {20, 40, 60};
+constexpr int kOpusSupportedFrameLengths[] = {10, 20, 40, 60};
+#endif
+
+// PacketLossFractionSmoother uses an exponential filter with a time constant
+// of -1.0 / ln(0.9999) = 10000 ms.
+constexpr float kAlphaForPacketLossFractionSmoother = 0.9999f;
+constexpr float kMaxPacketLossFraction = 0.2f;
+
+int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
+  const int bitrate = [&] {
+    if (max_playback_rate <= 8000) {
+      return kOpusBitrateNbBps * rtc::dchecked_cast<int>(num_channels);
+    } else if (max_playback_rate <= 16000) {
+      return kOpusBitrateWbBps * rtc::dchecked_cast<int>(num_channels);
+    } else {
+      return kOpusBitrateFbBps * rtc::dchecked_cast<int>(num_channels);
+    }
+  }();
+  RTC_DCHECK_GE(bitrate, AudioEncoderOpusConfig::kMinBitrateBps);
+  RTC_DCHECK_LE(bitrate, AudioEncoderOpusConfig::kMaxBitrateBps);
+  return bitrate;
+}
+
+// Get the maxaveragebitrate parameter in string-form, so we can properly figure
+// out how invalid it is and accurately log invalid values.
+int CalculateBitrate(int max_playback_rate_hz,
+                     size_t num_channels,
+                     absl::optional<std::string> bitrate_param) {
+  const int default_bitrate =
+      CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
+
+  if (bitrate_param) {
+    const auto bitrate = rtc::StringToNumber<int>(*bitrate_param);
+    if (bitrate) {
+      const int chosen_bitrate =
+          std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                   std::min(*bitrate, AudioEncoderOpusConfig::kMaxBitrateBps));
+      if (bitrate != chosen_bitrate) {
+        RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate " << *bitrate
+                            << " clamped to " << chosen_bitrate;
+      }
+      return chosen_bitrate;
+    }
+    RTC_LOG(LS_WARNING) << "Invalid maxaveragebitrate \"" << *bitrate_param
+                        << "\" replaced by default bitrate " << default_bitrate;
+  }
+
+  return default_bitrate;
+}
+
+int GetChannelCount(const SdpAudioFormat& format) {
+  const auto param = GetFormatParameter(format, "stereo");
+  if (param == "1") {
+    return 2;
+  } else {
+    return 1;
+  }
+}
+
+int GetMaxPlaybackRate(const SdpAudioFormat& format) {
+  const auto param = GetFormatParameter<int>(format, "maxplaybackrate");
+  if (param && *param >= 8000) {
+    return std::min(*param, kDefaultMaxPlaybackRate);
+  }
+  return kDefaultMaxPlaybackRate;
+}
+
+int GetFrameSizeMs(const SdpAudioFormat& format) {
+  const auto ptime = GetFormatParameter<int>(format, "ptime");
+  if (ptime) {
+    // Pick the next highest supported frame length from
+    // kOpusSupportedFrameLengths.
+    for (const int supported_frame_length : kOpusSupportedFrameLengths) {
+      if (supported_frame_length >= *ptime) {
+        return supported_frame_length;
+      }
+    }
+    // If none was found, return the largest supported frame length.
+    return *(std::end(kOpusSupportedFrameLengths) - 1);
+  }
+
+  return AudioEncoderOpusConfig::kDefaultFrameSizeMs;
+}
+
+void FindSupportedFrameLengths(int min_frame_length_ms,
+                               int max_frame_length_ms,
+                               std::vector<int>* out) {
+  out->clear();
+  std::copy_if(std::begin(kANASupportedFrameLengths),
+               std::end(kANASupportedFrameLengths), std::back_inserter(*out),
+               [&](int frame_length_ms) {
+                 return frame_length_ms >= min_frame_length_ms &&
+                        frame_length_ms <= max_frame_length_ms;
+               });
+  RTC_DCHECK(std::is_sorted(out->begin(), out->end()));
+}
+
+int GetBitrateBps(const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  return *config.bitrate_bps;
+}
+
+std::vector<float> GetBitrateMultipliers() {
+  constexpr char kBitrateMultipliersName[] =
+      "WebRTC-Audio-OpusBitrateMultipliers";
+  const bool use_bitrate_multipliers =
+      webrtc::field_trial::IsEnabled(kBitrateMultipliersName);
+  if (use_bitrate_multipliers) {
+    const std::string field_trial_string =
+        webrtc::field_trial::FindFullName(kBitrateMultipliersName);
+    std::vector<std::string> pieces;
+    rtc::tokenize(field_trial_string, '-', &pieces);
+    if (pieces.size() < 2 || pieces[0] != "Enabled") {
+      RTC_LOG(LS_WARNING) << "Invalid parameters for "
+                          << kBitrateMultipliersName
+                          << ", not using custom values.";
+      return std::vector<float>();
+    }
+    std::vector<float> multipliers(pieces.size() - 1);
+    for (size_t i = 1; i < pieces.size(); i++) {
+      if (!rtc::FromString(pieces[i], &multipliers[i - 1])) {
+        RTC_LOG(LS_WARNING)
+            << "Invalid parameters for " << kBitrateMultipliersName
+            << ", not using custom values.";
+        return std::vector<float>();
+      }
+    }
+    RTC_LOG(LS_INFO) << "Using custom bitrate multipliers: "
+                     << field_trial_string;
+    return multipliers;
+  }
+  return std::vector<float>();
+}
+
+int GetMultipliedBitrate(int bitrate, const std::vector<float>& multipliers) {
+  // The multipliers are valid from 5 kbps.
+  const size_t bitrate_kbps = static_cast<size_t>(bitrate / 1000);
+  if (bitrate_kbps < 5 || bitrate_kbps >= multipliers.size() + 5) {
+    return bitrate;
+  }
+  return static_cast<int>(multipliers[bitrate_kbps - 5] * bitrate);
+}
+}  // namespace
+
+void AudioEncoderOpusImpl::AppendSupportedEncoders(
+    std::vector<AudioCodecSpec>* specs) {
+  const SdpAudioFormat fmt = {"opus",
+                              kRtpTimestampRateHz,
+                              2,
+                              {{"minptime", "10"}, {"useinbandfec", "1"}}};
+  const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
+  specs->push_back({fmt, info});
+}
+
+AudioCodecInfo AudioEncoderOpusImpl::QueryAudioEncoder(
+    const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  AudioCodecInfo info(config.sample_rate_hz, config.num_channels,
+                      *config.bitrate_bps,
+                      AudioEncoderOpusConfig::kMinBitrateBps,
+                      AudioEncoderOpusConfig::kMaxBitrateBps);
+  info.allow_comfort_noise = false;
+  info.supports_network_adaption = true;
+  return info;
+}
+
+std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
+    const AudioEncoderOpusConfig& config,
+    int payload_type) {
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  return std::make_unique<AudioEncoderOpusImpl>(config, payload_type);
+}
+
+absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
+    const SdpAudioFormat& format) {
+  if (!absl::EqualsIgnoreCase(format.name, "opus") ||
+      format.clockrate_hz != kRtpTimestampRateHz) {
+    return absl::nullopt;
+  }
+
+  AudioEncoderOpusConfig config;
+  config.num_channels = GetChannelCount(format);
+  config.frame_size_ms = GetFrameSizeMs(format);
+  config.max_playback_rate_hz = GetMaxPlaybackRate(format);
+  config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
+  config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
+  config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
+  config.bitrate_bps =
+      CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
+                       GetFormatParameter(format, "maxaveragebitrate"));
+  config.application = config.num_channels == 1
+                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
+
+  constexpr int kMinANAFrameLength = kANASupportedFrameLengths[0];
+  constexpr int kMaxANAFrameLength =
+      kANASupportedFrameLengths[arraysize(kANASupportedFrameLengths) - 1];
+
+  // For now, minptime and maxptime are only used with ANA. If ptime is outside
+  // of this range, it will get adjusted once ANA takes hold. Ideally, we'd know
+  // if ANA was to be used when setting up the config, and adjust accordingly.
+  const int min_frame_length_ms =
+      GetFormatParameter<int>(format, "minptime").value_or(kMinANAFrameLength);
+  const int max_frame_length_ms =
+      GetFormatParameter<int>(format, "maxptime").value_or(kMaxANAFrameLength);
+
+  FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
+                            &config.supported_frame_lengths_ms);
+  if (!config.IsOk()) {
+    RTC_DCHECK_NOTREACHED();
+    return absl::nullopt;
+  }
+  return config;
+}
+
+absl::optional<int> AudioEncoderOpusImpl::GetNewComplexity(
+    const AudioEncoderOpusConfig& config) {
+  RTC_DCHECK(config.IsOk());
+  const int bitrate_bps = GetBitrateBps(config);
+  if (bitrate_bps >= config.complexity_threshold_bps -
+                         config.complexity_threshold_window_bps &&
+      bitrate_bps <= config.complexity_threshold_bps +
+                         config.complexity_threshold_window_bps) {
+    // Within the hysteresis window; make no change.
+    return absl::nullopt;
+  } else {
+    return bitrate_bps <= config.complexity_threshold_bps
+               ? config.low_rate_complexity
+               : config.complexity;
+  }
+}
+
+absl::optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
+    const AudioEncoderOpusConfig& config,
+    OpusEncInst* inst) {
+  constexpr int kMinWidebandBitrate = 8000;
+  constexpr int kMaxNarrowbandBitrate = 9000;
+  constexpr int kAutomaticThreshold = 11000;
+  RTC_DCHECK(config.IsOk());
+  const int bitrate = GetBitrateBps(config);
+  if (bitrate > kAutomaticThreshold) {
+    return absl::optional<int>(OPUS_AUTO);
+  }
+  const int bandwidth = WebRtcOpus_GetBandwidth(inst);
+  RTC_DCHECK_GE(bandwidth, 0);
+  if (bitrate > kMaxNarrowbandBitrate && bandwidth < OPUS_BANDWIDTH_WIDEBAND) {
+    return absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND);
+  } else if (bitrate < kMinWidebandBitrate &&
+             bandwidth > OPUS_BANDWIDTH_NARROWBAND) {
+    return absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND);
+  }
+  return absl::optional<int>();
+}
+
+class AudioEncoderOpusImpl::PacketLossFractionSmoother {
+ public:
+  explicit PacketLossFractionSmoother()
+      : last_sample_time_ms_(rtc::TimeMillis()),
+        smoother_(kAlphaForPacketLossFractionSmoother) {}
+
+  // Gets the smoothed packet loss fraction.
+  float GetAverage() const {
+    float value = smoother_.filtered();
+    return (value == rtc::ExpFilter::kValueUndefined) ? 0.0f : value;
+  }
+
+  // Add new observation to the packet loss fraction smoother.
+  void AddSample(float packet_loss_fraction) {
+    int64_t now_ms = rtc::TimeMillis();
+    smoother_.Apply(static_cast<float>(now_ms - last_sample_time_ms_),
+                    packet_loss_fraction);
+    last_sample_time_ms_ = now_ms;
+  }
+
+ private:
+  int64_t last_sample_time_ms_;
+
+  // An exponential filter is used to smooth the packet loss fraction.
+  rtc::ExpFilter smoother_;
+};
+
+AudioEncoderOpusImpl::AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config,
+                                           int payload_type)
+    : AudioEncoderOpusImpl(
+          config,
+          payload_type,
+          [this](absl::string_view config_string, RtcEventLog* event_log) {
+            return DefaultAudioNetworkAdaptorCreator(config_string, event_log);
+          },
+          // We choose 5sec as initial time constant due to empirical data.
+          std::make_unique<SmoothingFilterImpl>(5000)) {}
+
+AudioEncoderOpusImpl::AudioEncoderOpusImpl(
+    const AudioEncoderOpusConfig& config,
+    int payload_type,
+    const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
+    std::unique_ptr<SmoothingFilter> bitrate_smoother)
+    : payload_type_(payload_type),
+      use_stable_target_for_adaptation_(!webrtc::field_trial::IsDisabled(
+          "WebRTC-Audio-StableTargetAdaptation")),
+      adjust_bandwidth_(
+          webrtc::field_trial::IsEnabled("WebRTC-AdjustOpusBandwidth")),
+      bitrate_changed_(true),
+      bitrate_multipliers_(GetBitrateMultipliers()),
+      packet_loss_rate_(0.0),
+      inst_(nullptr),
+      packet_loss_fraction_smoother_(new PacketLossFractionSmoother()),
+      audio_network_adaptor_creator_(audio_network_adaptor_creator),
+      bitrate_smoother_(std::move(bitrate_smoother)),
+      consecutive_dtx_frames_(0) {
+  RTC_DCHECK(0 <= payload_type && payload_type <= 127);
+
+  // Sanity check of the redundant payload type field that we want to get rid
+  // of. See https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+  RTC_CHECK(config.payload_type == -1 || config.payload_type == payload_type);
+
+  RTC_CHECK(RecreateEncoderInstance(config));
+  SetProjectedPacketLossRate(packet_loss_rate_);
+}
+
+AudioEncoderOpusImpl::AudioEncoderOpusImpl(int payload_type,
+                                           const SdpAudioFormat& format)
+    : AudioEncoderOpusImpl(*SdpToConfig(format), payload_type) {}
+
+AudioEncoderOpusImpl::~AudioEncoderOpusImpl() {
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+}
+
+int AudioEncoderOpusImpl::SampleRateHz() const {
+  return config_.sample_rate_hz;
+}
+
+size_t AudioEncoderOpusImpl::NumChannels() const {
+  return config_.num_channels;
+}
+
+int AudioEncoderOpusImpl::RtpTimestampRateHz() const {
+  return kRtpTimestampRateHz;
+}
+
+size_t AudioEncoderOpusImpl::Num10MsFramesInNextPacket() const {
+  return Num10msFramesPerPacket();
+}
+
+size_t AudioEncoderOpusImpl::Max10MsFramesInAPacket() const {
+  return Num10msFramesPerPacket();
+}
+
+int AudioEncoderOpusImpl::GetTargetBitrate() const {
+  return GetBitrateBps(config_);
+}
+
+void AudioEncoderOpusImpl::Reset() {
+  RTC_CHECK(RecreateEncoderInstance(config_));
+}
+
+bool AudioEncoderOpusImpl::SetFec(bool enable) {
+  if (enable) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+  }
+  config_.fec_enabled = enable;
+  return true;
+}
+
+bool AudioEncoderOpusImpl::SetDtx(bool enable) {
+  if (enable) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+  }
+  config_.dtx_enabled = enable;
+  return true;
+}
+
+bool AudioEncoderOpusImpl::GetDtx() const {
+  return config_.dtx_enabled;
+}
+
+bool AudioEncoderOpusImpl::SetApplication(Application application) {
+  auto conf = config_;
+  switch (application) {
+    case Application::kSpeech:
+      conf.application = AudioEncoderOpusConfig::ApplicationMode::kVoip;
+      break;
+    case Application::kAudio:
+      conf.application = AudioEncoderOpusConfig::ApplicationMode::kAudio;
+      break;
+  }
+  return RecreateEncoderInstance(conf);
+}
+
+void AudioEncoderOpusImpl::SetMaxPlaybackRate(int frequency_hz) {
+  auto conf = config_;
+  conf.max_playback_rate_hz = frequency_hz;
+  RTC_CHECK(RecreateEncoderInstance(conf));
+}
+
+bool AudioEncoderOpusImpl::EnableAudioNetworkAdaptor(
+    const std::string& config_string,
+    RtcEventLog* event_log) {
+  audio_network_adaptor_ =
+      audio_network_adaptor_creator_(config_string, event_log);
+  return audio_network_adaptor_.get() != nullptr;
+}
+
+void AudioEncoderOpusImpl::DisableAudioNetworkAdaptor() {
+  audio_network_adaptor_.reset(nullptr);
+}
+
+void AudioEncoderOpusImpl::OnReceivedUplinkPacketLossFraction(
+    float uplink_packet_loss_fraction) {
+  if (audio_network_adaptor_) {
+    audio_network_adaptor_->SetUplinkPacketLossFraction(
+        uplink_packet_loss_fraction);
+    ApplyAudioNetworkAdaptor();
+  }
+  packet_loss_fraction_smoother_->AddSample(uplink_packet_loss_fraction);
+  float average_fraction_loss = packet_loss_fraction_smoother_->GetAverage();
+  SetProjectedPacketLossRate(average_fraction_loss);
+}
+
+void AudioEncoderOpusImpl::OnReceivedTargetAudioBitrate(
+    int target_audio_bitrate_bps) {
+  SetTargetBitrate(target_audio_bitrate_bps);
+}
+
+void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms,
+    absl::optional<int64_t> stable_target_bitrate_bps) {
+  if (audio_network_adaptor_) {
+    audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
+    if (use_stable_target_for_adaptation_) {
+      if (stable_target_bitrate_bps)
+        audio_network_adaptor_->SetUplinkBandwidth(*stable_target_bitrate_bps);
+    } else {
+      // We give smoothed bitrate allocation to audio network adaptor as
+      // the uplink bandwidth.
+      // The BWE spikes should not affect the bitrate smoother more than 25%.
+      // To simplify the calculations we use a step response as input signal.
+      // The step response of an exponential filter is
+      // u(t) = 1 - e^(-t / time_constant).
+      // In order to limit the affect of a BWE spike within 25% of its value
+      // before
+      // the next BWE update, we would choose a time constant that fulfills
+      // 1 - e^(-bwe_period_ms / time_constant) < 0.25
+      // Then 4 * bwe_period_ms is a good choice.
+      if (bwe_period_ms)
+        bitrate_smoother_->SetTimeConstantMs(*bwe_period_ms * 4);
+      bitrate_smoother_->AddSample(target_audio_bitrate_bps);
+    }
+
+    ApplyAudioNetworkAdaptor();
+  } else {
+    if (!overhead_bytes_per_packet_) {
+      RTC_LOG(LS_INFO)
+          << "AudioEncoderOpusImpl: Overhead unknown, target audio bitrate "
+          << target_audio_bitrate_bps << " bps is ignored.";
+      return;
+    }
+    const int overhead_bps = static_cast<int>(
+        *overhead_bytes_per_packet_ * 8 * 100 / Num10MsFramesInNextPacket());
+    SetTargetBitrate(
+        std::min(AudioEncoderOpusConfig::kMaxBitrateBps,
+                 std::max(AudioEncoderOpusConfig::kMinBitrateBps,
+                          target_audio_bitrate_bps - overhead_bps)));
+  }
+}
+void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
+    int target_audio_bitrate_bps,
+    absl::optional<int64_t> bwe_period_ms) {
+  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, bwe_period_ms,
+                            absl::nullopt);
+}
+
+void AudioEncoderOpusImpl::OnReceivedUplinkAllocation(
+    BitrateAllocationUpdate update) {
+  OnReceivedUplinkBandwidth(update.target_bitrate.bps(), update.bwe_period.ms(),
+                            update.stable_target_bitrate.bps());
+}
+
+void AudioEncoderOpusImpl::OnReceivedRtt(int rtt_ms) {
+  if (!audio_network_adaptor_)
+    return;
+  audio_network_adaptor_->SetRtt(rtt_ms);
+  ApplyAudioNetworkAdaptor();
+}
+
+void AudioEncoderOpusImpl::OnReceivedOverhead(
+    size_t overhead_bytes_per_packet) {
+  if (audio_network_adaptor_) {
+    audio_network_adaptor_->SetOverhead(overhead_bytes_per_packet);
+    ApplyAudioNetworkAdaptor();
+  } else {
+    overhead_bytes_per_packet_ = overhead_bytes_per_packet;
+  }
+}
+
+void AudioEncoderOpusImpl::SetReceiverFrameLengthRange(
+    int min_frame_length_ms,
+    int max_frame_length_ms) {
+  // Ensure that `SetReceiverFrameLengthRange` is called before
+  // `EnableAudioNetworkAdaptor`, otherwise we need to recreate
+  // `audio_network_adaptor_`, which is not a needed use case.
+  RTC_DCHECK(!audio_network_adaptor_);
+  FindSupportedFrameLengths(min_frame_length_ms, max_frame_length_ms,
+                            &config_.supported_frame_lengths_ms);
+}
+
+AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
+    uint32_t rtp_timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  MaybeUpdateUplinkBandwidth();
+
+  if (input_buffer_.empty())
+    first_timestamp_in_buffer_ = rtp_timestamp;
+
+  input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
+  if (input_buffer_.size() <
+      (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
+    return EncodedInfo();
+  }
+  RTC_CHECK_EQ(input_buffer_.size(),
+               Num10msFramesPerPacket() * SamplesPer10msFrame());
+
+  const size_t max_encoded_bytes = SufficientOutputBufferSize();
+  EncodedInfo info;
+  info.encoded_bytes = encoded->AppendData(
+      max_encoded_bytes, [&](rtc::ArrayView<uint8_t> encoded) {
+        int status = WebRtcOpus_Encode(
+            inst_, &input_buffer_[0],
+            rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
+            rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded.data());
+
+        RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
+
+        return static_cast<size_t>(status);
+      });
+  input_buffer_.clear();
+
+  bool dtx_frame = (info.encoded_bytes <= 2);
+
+  // Will use new packet size for next encoding.
+  config_.frame_size_ms = next_frame_length_ms_;
+
+  if (adjust_bandwidth_ && bitrate_changed_) {
+    const auto bandwidth = GetNewBandwidth(config_, inst_);
+    if (bandwidth) {
+      RTC_CHECK_EQ(0, WebRtcOpus_SetBandwidth(inst_, *bandwidth));
+    }
+    bitrate_changed_ = false;
+  }
+
+  info.encoded_timestamp = first_timestamp_in_buffer_;
+  info.payload_type = payload_type_;
+  info.send_even_if_empty = true;  // Allows Opus to send empty packets.
+  // After 20 DTX frames (MAX_CONSECUTIVE_DTX) Opus will send a frame
+  // coding the background noise. Avoid flagging this frame as speech
+  // (even though there is a probability of the frame being speech).
+  info.speech = !dtx_frame && (consecutive_dtx_frames_ != 20);
+  info.encoder_type = CodecType::kOpus;
+
+  // Increase or reset DTX counter.
+  consecutive_dtx_frames_ = (dtx_frame) ? (consecutive_dtx_frames_ + 1) : (0);
+
+  return info;
+}
+
+size_t AudioEncoderOpusImpl::Num10msFramesPerPacket() const {
+  return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
+}
+
+size_t AudioEncoderOpusImpl::SamplesPer10msFrame() const {
+  return rtc::CheckedDivExact(config_.sample_rate_hz, 100) *
+         config_.num_channels;
+}
+
+size_t AudioEncoderOpusImpl::SufficientOutputBufferSize() const {
+  // Calculate the number of bytes we expect the encoder to produce,
+  // then multiply by two to give a wide margin for error.
+  const size_t bytes_per_millisecond =
+      static_cast<size_t>(GetBitrateBps(config_) / (1000 * 8) + 1);
+  const size_t approx_encoded_bytes =
+      Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
+  return 2 * approx_encoded_bytes;
+}
+
+// If the given config is OK, recreate the Opus encoder instance with those
+// settings, save the config, and return true. Otherwise, do nothing and return
+// false.
+bool AudioEncoderOpusImpl::RecreateEncoderInstance(
+    const AudioEncoderOpusConfig& config) {
+  if (!config.IsOk())
+    return false;
+  config_ = config;
+  if (inst_)
+    RTC_CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
+  input_buffer_.clear();
+  input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame());
+  RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(
+                      &inst_, config.num_channels,
+                      config.application ==
+                              AudioEncoderOpusConfig::ApplicationMode::kVoip
+                          ? 0
+                          : 1,
+                      config.sample_rate_hz));
+  const int bitrate = GetBitrateBps(config);
+  RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, bitrate));
+  RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
+  if (config.fec_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
+  }
+  RTC_CHECK_EQ(
+      0, WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
+  // Use the default complexity if the start bitrate is within the hysteresis
+  // window.
+  complexity_ = GetNewComplexity(config).value_or(config.complexity);
+  RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
+  bitrate_changed_ = true;
+  if (config.dtx_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
+  }
+  RTC_CHECK_EQ(0,
+               WebRtcOpus_SetPacketLossRate(
+                   inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  if (config.cbr_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableCbr(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableCbr(inst_));
+  }
+  num_channels_to_encode_ = NumChannels();
+  next_frame_length_ms_ = config_.frame_size_ms;
+  return true;
+}
+
+void AudioEncoderOpusImpl::SetFrameLength(int frame_length_ms) {
+  if (next_frame_length_ms_ != frame_length_ms) {
+    RTC_LOG(LS_VERBOSE) << "Update Opus frame length "
+                        << "from " << next_frame_length_ms_ << " ms "
+                        << "to " << frame_length_ms << " ms.";
+  }
+  next_frame_length_ms_ = frame_length_ms;
+}
+
+void AudioEncoderOpusImpl::SetNumChannelsToEncode(
+    size_t num_channels_to_encode) {
+  RTC_DCHECK_GT(num_channels_to_encode, 0);
+  RTC_DCHECK_LE(num_channels_to_encode, config_.num_channels);
+
+  if (num_channels_to_encode_ == num_channels_to_encode)
+    return;
+
+  RTC_CHECK_EQ(0, WebRtcOpus_SetForceChannels(inst_, num_channels_to_encode));
+  num_channels_to_encode_ = num_channels_to_encode;
+}
+
+void AudioEncoderOpusImpl::SetProjectedPacketLossRate(float fraction) {
+  fraction = std::min(std::max(fraction, 0.0f), kMaxPacketLossFraction);
+  if (packet_loss_rate_ != fraction) {
+    packet_loss_rate_ = fraction;
+    RTC_CHECK_EQ(
+        0, WebRtcOpus_SetPacketLossRate(
+               inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
+  }
+}
+
+void AudioEncoderOpusImpl::SetTargetBitrate(int bits_per_second) {
+  const int new_bitrate = rtc::SafeClamp<int>(
+      bits_per_second, AudioEncoderOpusConfig::kMinBitrateBps,
+      AudioEncoderOpusConfig::kMaxBitrateBps);
+  if (config_.bitrate_bps && *config_.bitrate_bps != new_bitrate) {
+    config_.bitrate_bps = new_bitrate;
+    RTC_DCHECK(config_.IsOk());
+    const int bitrate = GetBitrateBps(config_);
+    RTC_CHECK_EQ(
+        0, WebRtcOpus_SetBitRate(
+               inst_, GetMultipliedBitrate(bitrate, bitrate_multipliers_)));
+    RTC_LOG(LS_VERBOSE) << "Set Opus bitrate to " << bitrate << " bps.";
+    bitrate_changed_ = true;
+  }
+
+  const auto new_complexity = GetNewComplexity(config_);
+  if (new_complexity && complexity_ != *new_complexity) {
+    complexity_ = *new_complexity;
+    RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
+  }
+}
+
+void AudioEncoderOpusImpl::ApplyAudioNetworkAdaptor() {
+  auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
+
+  if (config.bitrate_bps)
+    SetTargetBitrate(*config.bitrate_bps);
+  if (config.frame_length_ms)
+    SetFrameLength(*config.frame_length_ms);
+  if (config.enable_dtx)
+    SetDtx(*config.enable_dtx);
+  if (config.num_channels)
+    SetNumChannelsToEncode(*config.num_channels);
+}
+
+std::unique_ptr<AudioNetworkAdaptor>
+AudioEncoderOpusImpl::DefaultAudioNetworkAdaptorCreator(
+    absl::string_view config_string,
+    RtcEventLog* event_log) const {
+  AudioNetworkAdaptorImpl::Config config;
+  config.event_log = event_log;
+  return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
+      config, ControllerManagerImpl::Create(
+                  config_string, NumChannels(), supported_frame_lengths_ms(),
+                  AudioEncoderOpusConfig::kMinBitrateBps,
+                  num_channels_to_encode_, next_frame_length_ms_,
+                  GetTargetBitrate(), config_.fec_enabled, GetDtx())));
+}
+
+void AudioEncoderOpusImpl::MaybeUpdateUplinkBandwidth() {
+  if (audio_network_adaptor_ && !use_stable_target_for_adaptation_) {
+    int64_t now_ms = rtc::TimeMillis();
+    if (!bitrate_smoother_last_update_time_ ||
+        now_ms - *bitrate_smoother_last_update_time_ >=
+            config_.uplink_bandwidth_update_interval_ms) {
+      absl::optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
+      if (smoothed_bitrate)
+        audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
+      bitrate_smoother_last_update_time_ = now_ms;
+    }
+  }
+}
+
+ANAStats AudioEncoderOpusImpl::GetANAStats() const {
+  if (audio_network_adaptor_) {
+    return audio_network_adaptor_->GetStats();
+  }
+  return ANAStats();
+}
+
+absl::optional<std::pair<TimeDelta, TimeDelta> >
+AudioEncoderOpusImpl::GetFrameLengthRange() const {
+  if (audio_network_adaptor_) {
+    if (config_.supported_frame_lengths_ms.empty()) {
+      return absl::nullopt;
+    }
+    return {{TimeDelta::Millis(config_.supported_frame_lengths_ms.front()),
+             TimeDelta::Millis(config_.supported_frame_lengths_ms.back())}};
+  } else {
+    return {{TimeDelta::Millis(config_.frame_size_ms),
+             TimeDelta::Millis(config_.frame_size_ms)}};
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
new file mode 100644
index 0000000000..8c5c235016
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+
+#include <functional>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "common_audio/smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class AudioEncoderOpusImpl final : public AudioEncoder {
+ public:
+  // Returns empty if the current bitrate falls within the hysteresis window,
+  // defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
+  // Otherwise, returns the current complexity depending on whether the
+  // current bitrate is above or below complexity_threshold_bps.
+  static absl::optional<int> GetNewComplexity(
+      const AudioEncoderOpusConfig& config);
+
+  // Returns OPUS_AUTO if the the current bitrate is above wideband threshold.
+  // Returns empty if it is below, but bandwidth coincides with the desired one.
+  // Otherwise returns the desired bandwidth.
+  static absl::optional<int> GetNewBandwidth(
+      const AudioEncoderOpusConfig& config,
+      OpusEncInst* inst);
+
+  using AudioNetworkAdaptorCreator =
+      std::function<std::unique_ptr<AudioNetworkAdaptor>(absl::string_view,
+                                                         RtcEventLog*)>;
+
+  AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config, int payload_type);
+
+  // Dependency injection for testing.
+  AudioEncoderOpusImpl(
+      const AudioEncoderOpusConfig& config,
+      int payload_type,
+      const AudioNetworkAdaptorCreator& audio_network_adaptor_creator,
+      std::unique_ptr<SmoothingFilter> bitrate_smoother);
+
+  AudioEncoderOpusImpl(int payload_type, const SdpAudioFormat& format);
+  ~AudioEncoderOpusImpl() override;
+
+  AudioEncoderOpusImpl(const AudioEncoderOpusImpl&) = delete;
+  AudioEncoderOpusImpl& operator=(const AudioEncoderOpusImpl&) = delete;
+
+  int SampleRateHz() const override;
+  size_t NumChannels() const override;
+  int RtpTimestampRateHz() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
+  int GetTargetBitrate() const override;
+
+  void Reset() override;
+  bool SetFec(bool enable) override;
+
+  // Set Opus DTX. Once enabled, Opus stops transmission, when it detects
+  // voice being inactive. During that, it still sends 2 packets (one for
+  // content, one for signaling) about every 400 ms.
+  bool SetDtx(bool enable) override;
+  bool GetDtx() const override;
+
+  bool SetApplication(Application application) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
+  bool EnableAudioNetworkAdaptor(const std::string& config_string,
+                                 RtcEventLog* event_log) override;
+  void DisableAudioNetworkAdaptor() override;
+  void OnReceivedUplinkPacketLossFraction(
+      float uplink_packet_loss_fraction) override;
+  void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms) override;
+  void OnReceivedUplinkAllocation(BitrateAllocationUpdate update) override;
+  void OnReceivedRtt(int rtt_ms) override;
+  void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
+  void SetReceiverFrameLengthRange(int min_frame_length_ms,
+                                   int max_frame_length_ms) override;
+  ANAStats GetANAStats() const override;
+  absl::optional<std::pair<TimeDelta, TimeDelta> > GetFrameLengthRange()
+      const override;
+  rtc::ArrayView<const int> supported_frame_lengths_ms() const {
+    return config_.supported_frame_lengths_ms;
+  }
+
+  // Getters for testing.
+  float packet_loss_rate() const { return packet_loss_rate_; }
+  AudioEncoderOpusConfig::ApplicationMode application() const {
+    return config_.application;
+  }
+  bool fec_enabled() const { return config_.fec_enabled; }
+  size_t num_channels_to_encode() const { return num_channels_to_encode_; }
+  int next_frame_length_ms() const { return next_frame_length_ms_; }
+
+ protected:
+  EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
+                         rtc::ArrayView<const int16_t> audio,
+                         rtc::Buffer* encoded) override;
+
+ private:
+  class PacketLossFractionSmoother;
+
+  static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderOpusConfig&,
+      int payload_type);
+
+  size_t Num10msFramesPerPacket() const;
+  size_t SamplesPer10msFrame() const;
+  size_t SufficientOutputBufferSize() const;
+  bool RecreateEncoderInstance(const AudioEncoderOpusConfig& config);
+  void SetFrameLength(int frame_length_ms);
+  void SetNumChannelsToEncode(size_t num_channels_to_encode);
+  void SetProjectedPacketLossRate(float fraction);
+
+  void OnReceivedUplinkBandwidth(
+      int target_audio_bitrate_bps,
+      absl::optional<int64_t> bwe_period_ms,
+      absl::optional<int64_t> link_capacity_allocation);
+
+  // TODO(minyue): remove "override" when we can deprecate
+  // `AudioEncoder::SetTargetBitrate`.
+  void SetTargetBitrate(int target_bps) override;
+
+  void ApplyAudioNetworkAdaptor();
+  std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
+      absl::string_view config_string,
+      RtcEventLog* event_log) const;
+
+  void MaybeUpdateUplinkBandwidth();
+
+  AudioEncoderOpusConfig config_;
+  const int payload_type_;
+  const bool use_stable_target_for_adaptation_;
+  const bool adjust_bandwidth_;
+  bool bitrate_changed_;
+  // A multiplier for bitrates at 5 kbps and higher. The target bitrate
+  // will be multiplied by these multipliers, each multiplier is applied to a
+  // 1 kbps range.
+  std::vector<float> bitrate_multipliers_;
+  float packet_loss_rate_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
+  size_t num_channels_to_encode_;
+  int next_frame_length_ms_;
+  int complexity_;
+  std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
+  const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
+  std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
+  absl::optional<size_t> overhead_bytes_per_packet_;
+  const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
+  absl::optional<int64_t> bitrate_smoother_last_update_time_;
+  int consecutive_dtx_frames_;
+
+  friend struct AudioEncoderOpus;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
new file mode 100644
index 0000000000..a2ebe43bbe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -0,0 +1,914 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+
+#include <array>
+#include <memory>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "common_audio/mocks/mock_smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/fake_clock.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace {
+
+constexpr int kDefaultOpusPayloadType = 105;
+constexpr int kDefaultOpusRate = 32000;
+constexpr int kDefaultOpusPacSize = 960;
+constexpr int64_t kInitialTimeUs = 12345678;
+
+AudioEncoderOpusConfig CreateConfigWithParameters(
+    const SdpAudioFormat::Parameters& params) {
+  const SdpAudioFormat format("opus", 48000, 2, params);
+  return *AudioEncoderOpus::SdpToConfig(format);
+}
+
+struct AudioEncoderOpusStates {
+  MockAudioNetworkAdaptor* mock_audio_network_adaptor;
+  MockSmoothingFilter* mock_bitrate_smoother;
+  std::unique_ptr<AudioEncoderOpusImpl> encoder;
+  std::unique_ptr<rtc::ScopedFakeClock> fake_clock;
+  AudioEncoderOpusConfig config;
+};
+
+std::unique_ptr<AudioEncoderOpusStates> CreateCodec(int sample_rate_hz,
+                                                    size_t num_channels) {
+  std::unique_ptr<AudioEncoderOpusStates> states =
+      std::make_unique<AudioEncoderOpusStates>();
+  states->mock_audio_network_adaptor = nullptr;
+  states->fake_clock.reset(new rtc::ScopedFakeClock());
+  states->fake_clock->SetTime(Timestamp::Micros(kInitialTimeUs));
+
+  MockAudioNetworkAdaptor** mock_ptr = &states->mock_audio_network_adaptor;
+  AudioEncoderOpusImpl::AudioNetworkAdaptorCreator creator =
+      [mock_ptr](absl::string_view, RtcEventLog* event_log) {
+        std::unique_ptr<MockAudioNetworkAdaptor> adaptor(
+            new NiceMock<MockAudioNetworkAdaptor>());
+        EXPECT_CALL(*adaptor, Die());
+        *mock_ptr = adaptor.get();
+        return adaptor;
+      };
+
+  AudioEncoderOpusConfig config;
+  config.frame_size_ms = rtc::CheckedDivExact(kDefaultOpusPacSize, 48);
+  config.sample_rate_hz = sample_rate_hz;
+  config.num_channels = num_channels;
+  config.bitrate_bps = kDefaultOpusRate;
+  config.application = num_channels == 1
+                           ? AudioEncoderOpusConfig::ApplicationMode::kVoip
+                           : AudioEncoderOpusConfig::ApplicationMode::kAudio;
+  config.supported_frame_lengths_ms.push_back(config.frame_size_ms);
+  states->config = config;
+
+  std::unique_ptr<MockSmoothingFilter> bitrate_smoother(
+      new MockSmoothingFilter());
+  states->mock_bitrate_smoother = bitrate_smoother.get();
+
+  states->encoder.reset(
+      new AudioEncoderOpusImpl(states->config, kDefaultOpusPayloadType, creator,
+                               std::move(bitrate_smoother)));
+  return states;
+}
+
+AudioEncoderRuntimeConfig CreateEncoderRuntimeConfig() {
+  constexpr int kBitrate = 40000;
+  constexpr int kFrameLength = 60;
+  constexpr bool kEnableDtx = false;
+  constexpr size_t kNumChannels = 1;
+  AudioEncoderRuntimeConfig config;
+  config.bitrate_bps = kBitrate;
+  config.frame_length_ms = kFrameLength;
+  config.enable_dtx = kEnableDtx;
+  config.num_channels = kNumChannels;
+  return config;
+}
+
+void CheckEncoderRuntimeConfig(const AudioEncoderOpusImpl* encoder,
+                               const AudioEncoderRuntimeConfig& config) {
+  EXPECT_EQ(*config.bitrate_bps, encoder->GetTargetBitrate());
+  EXPECT_EQ(*config.frame_length_ms, encoder->next_frame_length_ms());
+  EXPECT_EQ(*config.enable_dtx, encoder->GetDtx());
+  EXPECT_EQ(*config.num_channels, encoder->num_channels_to_encode());
+}
+
+// Create 10ms audio data blocks for a total packet size of "packet_size_ms".
+std::unique_ptr<test::AudioLoop> Create10msAudioBlocks(
+    const std::unique_ptr<AudioEncoderOpusImpl>& encoder,
+    int packet_size_ms) {
+  const std::string file_name =
+      test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+
+  std::unique_ptr<test::AudioLoop> speech_data(new test::AudioLoop());
+  int audio_samples_per_ms =
+      rtc::CheckedDivExact(encoder->SampleRateHz(), 1000);
+  if (!speech_data->Init(
+          file_name,
+          packet_size_ms * audio_samples_per_ms *
+              encoder->num_channels_to_encode(),
+          10 * audio_samples_per_ms * encoder->num_channels_to_encode()))
+    return nullptr;
+  return speech_data;
+}
+
+}  // namespace
+
+class AudioEncoderOpusTest : public ::testing::TestWithParam<int> {
+ protected:
+  int sample_rate_hz_{GetParam()};
+};
+INSTANTIATE_TEST_SUITE_P(Param,
+                         AudioEncoderOpusTest,
+                         ::testing::Values(16000, 48000));
+
+TEST_P(AudioEncoderOpusTest, DefaultApplicationModeMono) {
+  auto states = CreateCodec(sample_rate_hz_, 1);
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, ChangeApplicationMode) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  EXPECT_TRUE(
+      states->encoder->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  // Trigger a reset.
+  states->encoder->Reset();
+  // Verify that the mode is still kAudio.
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
+            states->encoder->application());
+
+  // Now change to kVoip.
+  EXPECT_TRUE(
+      states->encoder->SetApplication(AudioEncoder::Application::kSpeech));
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+
+  // Trigger a reset again.
+  states->encoder->Reset();
+  // Verify that the mode is still kVoip.
+  EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
+            states->encoder->application());
+}
+
+TEST_P(AudioEncoderOpusTest, ToggleDtx) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  // Enable DTX
+  EXPECT_TRUE(states->encoder->SetDtx(true));
+  EXPECT_TRUE(states->encoder->GetDtx());
+  // Turn off DTX.
+  EXPECT_TRUE(states->encoder->SetDtx(false));
+  EXPECT_FALSE(states->encoder->GetDtx());
+}
+
+TEST_P(AudioEncoderOpusTest,
+       OnReceivedUplinkBandwidthWithoutAudioNetworkAdaptor) {
+  auto states = CreateCodec(sample_rate_hz_, 1);
+  // Constants are replicated from audio_states->encoderopus.cc.
+  const int kMinBitrateBps = 6000;
+  const int kMaxBitrateBps = 510000;
+  const int kOverheadBytesPerPacket = 64;
+  states->encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
+  const int kOverheadBps = 8 * kOverheadBytesPerPacket *
+                           rtc::CheckedDivExact(48000, kDefaultOpusPacSize);
+  // Set a too low bitrate.
+  states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps + kOverheadBps - 1,
+                                             absl::nullopt);
+  EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
+  // Set a too high bitrate.
+  states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + kOverheadBps + 1,
+                                             absl::nullopt);
+  EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
+  // Set the minimum rate.
+  states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps + kOverheadBps,
+                                             absl::nullopt);
+  EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
+  // Set the maximum rate.
+  states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + kOverheadBps,
+                                             absl::nullopt);
+  EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
+  // Set rates from kMaxBitrateBps up to 32000 bps.
+  for (int rate = kMinBitrateBps + kOverheadBps; rate <= 32000 + kOverheadBps;
+       rate += 1000) {
+    states->encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
+    EXPECT_EQ(rate - kOverheadBps, states->encoder->GetTargetBitrate());
+  }
+}
+
+TEST_P(AudioEncoderOpusTest, SetReceiverFrameLengthRange) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  // Before calling to `SetReceiverFrameLengthRange`,
+  // `supported_frame_lengths_ms` should contain only the frame length being
+  // used.
+  using ::testing::ElementsAre;
+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+              ElementsAre(states->encoder->next_frame_length_ms()));
+  states->encoder->SetReceiverFrameLengthRange(0, 12345);
+  states->encoder->SetReceiverFrameLengthRange(21, 60);
+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+              ElementsAre(40, 60));
+  states->encoder->SetReceiverFrameLengthRange(20, 59);
+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
+              ElementsAre(20, 40));
+}
+
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkPacketLossFraction) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any packet loss fraction is fine.
+  constexpr float kUplinkPacketLoss = 0.1f;
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetUplinkPacketLossFraction(kUplinkPacketLoss));
+  states->encoder->OnReceivedUplinkPacketLossFraction(kUplinkPacketLoss);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-StableTargetAdaptation/Disabled/");
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any target audio bitrate is fine.
+  constexpr int kTargetAudioBitrate = 30000;
+  constexpr int64_t kProbingIntervalMs = 3000;
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetTargetAudioBitrate(kTargetAudioBitrate));
+  EXPECT_CALL(*states->mock_bitrate_smoother,
+              SetTimeConstantMs(kProbingIntervalMs * 4));
+  EXPECT_CALL(*states->mock_bitrate_smoother, AddSample(kTargetAudioBitrate));
+  states->encoder->OnReceivedUplinkBandwidth(kTargetAudioBitrate,
+                                             kProbingIntervalMs);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkAllocation) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  BitrateAllocationUpdate update;
+  update.target_bitrate = DataRate::BitsPerSec(30000);
+  update.stable_target_bitrate = DataRate::BitsPerSec(20000);
+  update.bwe_period = TimeDelta::Millis(200);
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetTargetAudioBitrate(update.target_bitrate.bps()));
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetUplinkBandwidth(update.stable_target_bitrate.bps()));
+  states->encoder->OnReceivedUplinkAllocation(update);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedRtt) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any rtt is fine.
+  constexpr int kRtt = 30;
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetRtt(kRtt));
+  states->encoder->OnReceivedRtt(kRtt);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedOverhead) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  // Since using mock audio network adaptor, any overhead is fine.
+  constexpr size_t kOverhead = 64;
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetOverhead(kOverhead));
+  states->encoder->OnReceivedOverhead(kOverhead);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest,
+       PacketLossFractionSmoothedOnSetUplinkPacketLossFraction) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  // The values are carefully chosen so that if no smoothing is made, the test
+  // will fail.
+  constexpr float kPacketLossFraction_1 = 0.02f;
+  constexpr float kPacketLossFraction_2 = 0.198f;
+  // `kSecondSampleTimeMs` is chosen to ease the calculation since
+  // 0.9999 ^ 6931 = 0.5.
+  constexpr int64_t kSecondSampleTimeMs = 6931;
+
+  // First time, no filtering.
+  states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_1);
+  EXPECT_FLOAT_EQ(0.02f, states->encoder->packet_loss_rate());
+
+  states->fake_clock->AdvanceTime(TimeDelta::Millis(kSecondSampleTimeMs));
+  states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_2);
+
+  // Now the output of packet loss fraction smoother should be
+  // (0.02 + 0.198) / 2 = 0.109.
+  EXPECT_NEAR(0.109f, states->encoder->packet_loss_rate(), 0.001);
+}
+
+TEST_P(AudioEncoderOpusTest, PacketLossRateUpperBounded) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  states->encoder->OnReceivedUplinkPacketLossFraction(0.5);
+  EXPECT_FLOAT_EQ(0.2f, states->encoder->packet_loss_rate());
+}
+
+TEST_P(AudioEncoderOpusTest, DoNotInvokeSetTargetBitrateIfOverheadUnknown) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+
+  states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusRate * 2,
+                                             absl::nullopt);
+
+  // Since `OnReceivedOverhead` has not been called, the codec bitrate should
+  // not change.
+  EXPECT_EQ(kDefaultOpusRate, states->encoder->GetTargetBitrate());
+}
+
+// Verifies that the complexity adaptation in the config works as intended.
+TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
+  AudioEncoderOpusConfig config;
+  config.low_rate_complexity = 8;
+  config.complexity = 6;
+
+  // Bitrate within hysteresis window. Expect empty output.
+  config.bitrate_bps = 12500;
+  EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
+
+  // Bitrate below hysteresis window. Expect higher complexity.
+  config.bitrate_bps = 10999;
+  EXPECT_EQ(8, AudioEncoderOpusImpl::GetNewComplexity(config));
+
+  // Bitrate within hysteresis window. Expect empty output.
+  config.bitrate_bps = 12500;
+  EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
+
+  // Bitrate above hysteresis window. Expect lower complexity.
+  config.bitrate_bps = 14001;
+  EXPECT_EQ(6, AudioEncoderOpusImpl::GetNewComplexity(config));
+}
+
+// Verifies that the bandwidth adaptation in the config works as intended.
+TEST_P(AudioEncoderOpusTest, ConfigBandwidthAdaptation) {
+  AudioEncoderOpusConfig config;
+  const size_t opus_rate_khz = rtc::CheckedDivExact(sample_rate_hz_, 1000);
+  const std::vector<int16_t> silence(
+      opus_rate_khz * config.frame_size_ms * config.num_channels, 0);
+  constexpr size_t kMaxBytes = 1000;
+  uint8_t bitstream[kMaxBytes];
+
+  OpusEncInst* inst;
+  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(
+                   &inst, config.num_channels,
+                   config.application ==
+                           AudioEncoderOpusConfig::ApplicationMode::kVoip
+                       ? 0
+                       : 1,
+                   sample_rate_hz_));
+
+  // Bitrate below minmum wideband. Expect narrowband.
+  config.bitrate_bps = absl::optional<int>(7999);
+  auto bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND), bandwidth);
+  WebRtcOpus_SetBandwidth(inst, *bandwidth);
+  // It is necessary to encode here because Opus has some logic in the encoder
+  // that goes from the user-set bandwidth to the used and returned one.
+  WebRtcOpus_Encode(inst, silence.data(),
+                    rtc::CheckedDivExact(silence.size(), config.num_channels),
+                    kMaxBytes, bitstream);
+
+  // Bitrate not yet above maximum narrowband. Expect empty.
+  config.bitrate_bps = absl::optional<int>(9000);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(), bandwidth);
+
+  // Bitrate above maximum narrowband. Expect wideband.
+  config.bitrate_bps = absl::optional<int>(9001);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND), bandwidth);
+  WebRtcOpus_SetBandwidth(inst, *bandwidth);
+  // It is necessary to encode here because Opus has some logic in the encoder
+  // that goes from the user-set bandwidth to the used and returned one.
+  WebRtcOpus_Encode(inst, silence.data(),
+                    rtc::CheckedDivExact(silence.size(), config.num_channels),
+                    kMaxBytes, bitstream);
+
+  // Bitrate not yet below minimum wideband. Expect empty.
+  config.bitrate_bps = absl::optional<int>(8000);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(), bandwidth);
+
+  // Bitrate above automatic threshold. Expect automatic.
+  config.bitrate_bps = absl::optional<int>(12001);
+  bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
+  EXPECT_EQ(absl::optional<int>(OPUS_AUTO), bandwidth);
+
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(inst));
+}
+
+TEST_P(AudioEncoderOpusTest, EmptyConfigDoesNotAffectEncoderSettings) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig empty_config;
+
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config))
+      .WillOnce(Return(empty_config));
+
+  constexpr size_t kOverhead = 64;
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetOverhead(kOverhead))
+      .Times(2);
+  states->encoder->OnReceivedOverhead(kOverhead);
+  states->encoder->OnReceivedOverhead(kOverhead);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
+TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-StableTargetAdaptation/Disabled/");
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+  const size_t opus_rate_khz = rtc::CheckedDivExact(sample_rate_hz_, 1000);
+  const std::vector<int16_t> audio(opus_rate_khz * 10 * 2, 0);
+  rtc::Buffer encoded;
+  EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
+      .WillOnce(Return(50000));
+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(50000));
+  states->encoder->Encode(
+      0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+
+  // Repeat update uplink bandwidth tests.
+  for (int i = 0; i < 5; i++) {
+    // Don't update till it is time to update again.
+    states->fake_clock->AdvanceTime(TimeDelta::Millis(
+        states->config.uplink_bandwidth_update_interval_ms - 1));
+    states->encoder->Encode(
+        0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+
+    // Update when it is time to update.
+    EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
+        .WillOnce(Return(40000));
+    EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(40000));
+    states->fake_clock->AdvanceTime(TimeDelta::Millis(1));
+    states->encoder->Encode(
+        0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
+  }
+}
+
+TEST_P(AudioEncoderOpusTest, EncodeAtMinBitrate) {
+  auto states = CreateCodec(sample_rate_hz_, 1);
+  constexpr int kNumPacketsToEncode = 2;
+  auto audio_frames =
+      Create10msAudioBlocks(states->encoder, kNumPacketsToEncode * 20);
+  ASSERT_TRUE(audio_frames) << "Create10msAudioBlocks failed";
+  rtc::Buffer encoded;
+  uint32_t rtp_timestamp = 12345;  // Just a number not important to this test.
+
+  states->encoder->OnReceivedUplinkBandwidth(0, absl::nullopt);
+  for (int packet_index = 0; packet_index < kNumPacketsToEncode;
+       packet_index++) {
+    // Make sure we are not encoding before we have enough data for
+    // a 20ms packet.
+    for (int index = 0; index < 1; index++) {
+      states->encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
+                              &encoded);
+      EXPECT_EQ(0u, encoded.size());
+    }
+
+    // Should encode now.
+    states->encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
+                            &encoded);
+    EXPECT_GT(encoded.size(), 0u);
+    encoded.Clear();
+  }
+}
+
+TEST(AudioEncoderOpusTest, TestConfigDefaults) {
+  const auto config_opt = AudioEncoderOpus::SdpToConfig({"opus", 48000, 2});
+  ASSERT_TRUE(config_opt);
+  EXPECT_EQ(48000, config_opt->max_playback_rate_hz);
+  EXPECT_EQ(1u, config_opt->num_channels);
+  EXPECT_FALSE(config_opt->fec_enabled);
+  EXPECT_FALSE(config_opt->dtx_enabled);
+  EXPECT_EQ(20, config_opt->frame_size_ms);
+}
+
+TEST(AudioEncoderOpusTest, TestConfigFromParams) {
+  const auto config1 = CreateConfigWithParameters({{"stereo", "0"}});
+  EXPECT_EQ(1U, config1.num_channels);
+
+  const auto config2 = CreateConfigWithParameters({{"stereo", "1"}});
+  EXPECT_EQ(2U, config2.num_channels);
+
+  const auto config3 = CreateConfigWithParameters({{"useinbandfec", "0"}});
+  EXPECT_FALSE(config3.fec_enabled);
+
+  const auto config4 = CreateConfigWithParameters({{"useinbandfec", "1"}});
+  EXPECT_TRUE(config4.fec_enabled);
+
+  const auto config5 = CreateConfigWithParameters({{"usedtx", "0"}});
+  EXPECT_FALSE(config5.dtx_enabled);
+
+  const auto config6 = CreateConfigWithParameters({{"usedtx", "1"}});
+  EXPECT_TRUE(config6.dtx_enabled);
+
+  const auto config7 = CreateConfigWithParameters({{"cbr", "0"}});
+  EXPECT_FALSE(config7.cbr_enabled);
+
+  const auto config8 = CreateConfigWithParameters({{"cbr", "1"}});
+  EXPECT_TRUE(config8.cbr_enabled);
+
+  const auto config9 =
+      CreateConfigWithParameters({{"maxplaybackrate", "12345"}});
+  EXPECT_EQ(12345, config9.max_playback_rate_hz);
+
+  const auto config10 =
+      CreateConfigWithParameters({{"maxaveragebitrate", "96000"}});
+  EXPECT_EQ(96000, config10.bitrate_bps);
+
+  const auto config11 = CreateConfigWithParameters({{"maxptime", "40"}});
+  for (int frame_length : config11.supported_frame_lengths_ms) {
+    EXPECT_LE(frame_length, 40);
+  }
+
+  const auto config12 = CreateConfigWithParameters({{"minptime", "40"}});
+  for (int frame_length : config12.supported_frame_lengths_ms) {
+    EXPECT_GE(frame_length, 40);
+  }
+
+  const auto config13 = CreateConfigWithParameters({{"ptime", "40"}});
+  EXPECT_EQ(40, config13.frame_size_ms);
+
+  constexpr int kMinSupportedFrameLength = 10;
+  constexpr int kMaxSupportedFrameLength =
+      WEBRTC_OPUS_SUPPORT_120MS_PTIME ? 120 : 60;
+
+  const auto config14 = CreateConfigWithParameters({{"ptime", "1"}});
+  EXPECT_EQ(kMinSupportedFrameLength, config14.frame_size_ms);
+
+  const auto config15 = CreateConfigWithParameters({{"ptime", "2000"}});
+  EXPECT_EQ(kMaxSupportedFrameLength, config15.frame_size_ms);
+}
+
+TEST(AudioEncoderOpusTest, TestConfigFromInvalidParams) {
+  const webrtc::SdpAudioFormat format("opus", 48000, 2);
+  const auto default_config = *AudioEncoderOpus::SdpToConfig(format);
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+  const std::vector<int> default_supported_frame_lengths_ms({20, 40, 60, 120});
+#else
+  const std::vector<int> default_supported_frame_lengths_ms({20, 40, 60});
+#endif
+
+  AudioEncoderOpusConfig config;
+  config = CreateConfigWithParameters({{"stereo", "invalid"}});
+  EXPECT_EQ(default_config.num_channels, config.num_channels);
+
+  config = CreateConfigWithParameters({{"useinbandfec", "invalid"}});
+  EXPECT_EQ(default_config.fec_enabled, config.fec_enabled);
+
+  config = CreateConfigWithParameters({{"usedtx", "invalid"}});
+  EXPECT_EQ(default_config.dtx_enabled, config.dtx_enabled);
+
+  config = CreateConfigWithParameters({{"cbr", "invalid"}});
+  EXPECT_EQ(default_config.dtx_enabled, config.dtx_enabled);
+
+  config = CreateConfigWithParameters({{"maxplaybackrate", "0"}});
+  EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
+
+  config = CreateConfigWithParameters({{"maxplaybackrate", "-23"}});
+  EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
+
+  config = CreateConfigWithParameters({{"maxplaybackrate", "not a number!"}});
+  EXPECT_EQ(default_config.max_playback_rate_hz, config.max_playback_rate_hz);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "0"}});
+  EXPECT_EQ(6000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "-1000"}});
+  EXPECT_EQ(6000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "1024000"}});
+  EXPECT_EQ(510000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxaveragebitrate", "not a number!"}});
+  EXPECT_EQ(default_config.bitrate_bps, config.bitrate_bps);
+
+  config = CreateConfigWithParameters({{"maxptime", "invalid"}});
+  EXPECT_EQ(default_supported_frame_lengths_ms,
+            config.supported_frame_lengths_ms);
+
+  config = CreateConfigWithParameters({{"minptime", "invalid"}});
+  EXPECT_EQ(default_supported_frame_lengths_ms,
+            config.supported_frame_lengths_ms);
+
+  config = CreateConfigWithParameters({{"ptime", "invalid"}});
+  EXPECT_EQ(default_supported_frame_lengths_ms,
+            config.supported_frame_lengths_ms);
+}
+
+TEST(AudioEncoderOpusTest, GetFrameLenghtRange) {
+  AudioEncoderOpusConfig config =
+      CreateConfigWithParameters({{"maxptime", "10"}, {"ptime", "10"}});
+  std::unique_ptr<AudioEncoder> encoder =
+      AudioEncoderOpus::MakeAudioEncoder(config, kDefaultOpusPayloadType);
+  auto ptime = webrtc::TimeDelta::Millis(10);
+  absl::optional<std::pair<webrtc::TimeDelta, webrtc::TimeDelta>> range = {
+      {ptime, ptime}};
+  EXPECT_EQ(encoder->GetFrameLengthRange(), range);
+}
+
+// Test that bitrate will be overridden by the "maxaveragebitrate" parameter.
+// Also test that the "maxaveragebitrate" can't be set to values outside the
+// range of 6000 and 510000
+TEST(AudioEncoderOpusTest, SetSendCodecOpusMaxAverageBitrate) {
+  // Ignore if less than 6000.
+  const auto config1 = AudioEncoderOpus::SdpToConfig(
+      {"opus", 48000, 2, {{"maxaveragebitrate", "5999"}}});
+  EXPECT_EQ(6000, config1->bitrate_bps);
+
+  // Ignore if larger than 510000.
+  const auto config2 = AudioEncoderOpus::SdpToConfig(
+      {"opus", 48000, 2, {{"maxaveragebitrate", "510001"}}});
+  EXPECT_EQ(510000, config2->bitrate_bps);
+
+  const auto config3 = AudioEncoderOpus::SdpToConfig(
+      {"opus", 48000, 2, {{"maxaveragebitrate", "200000"}}});
+  EXPECT_EQ(200000, config3->bitrate_bps);
+}
+
+// Test maxplaybackrate <= 8000 triggers Opus narrow band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateNb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "8000"}});
+  EXPECT_EQ(8000, config.max_playback_rate_hz);
+  EXPECT_EQ(12000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "8000"}, {"stereo", "1"}});
+  EXPECT_EQ(8000, config.max_playback_rate_hz);
+  EXPECT_EQ(24000, config.bitrate_bps);
+}
+
+// Test 8000 < maxplaybackrate <= 12000 triggers Opus medium band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateMb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "8001"}});
+  EXPECT_EQ(8001, config.max_playback_rate_hz);
+  EXPECT_EQ(20000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "8001"}, {"stereo", "1"}});
+  EXPECT_EQ(8001, config.max_playback_rate_hz);
+  EXPECT_EQ(40000, config.bitrate_bps);
+}
+
+// Test 12000 < maxplaybackrate <= 16000 triggers Opus wide band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateWb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "12001"}});
+  EXPECT_EQ(12001, config.max_playback_rate_hz);
+  EXPECT_EQ(20000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "12001"}, {"stereo", "1"}});
+  EXPECT_EQ(12001, config.max_playback_rate_hz);
+  EXPECT_EQ(40000, config.bitrate_bps);
+}
+
+// Test 16000 < maxplaybackrate <= 24000 triggers Opus super wide band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateSwb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "16001"}});
+  EXPECT_EQ(16001, config.max_playback_rate_hz);
+  EXPECT_EQ(32000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "16001"}, {"stereo", "1"}});
+  EXPECT_EQ(16001, config.max_playback_rate_hz);
+  EXPECT_EQ(64000, config.bitrate_bps);
+}
+
+// Test 24000 < maxplaybackrate triggers Opus full band mode.
+TEST(AudioEncoderOpusTest, SetMaxPlaybackRateFb) {
+  auto config = CreateConfigWithParameters({{"maxplaybackrate", "24001"}});
+  EXPECT_EQ(24001, config.max_playback_rate_hz);
+  EXPECT_EQ(32000, config.bitrate_bps);
+
+  config = CreateConfigWithParameters(
+      {{"maxplaybackrate", "24001"}, {"stereo", "1"}});
+  EXPECT_EQ(24001, config.max_playback_rate_hz);
+  EXPECT_EQ(64000, config.bitrate_bps);
+}
+
+TEST_P(AudioEncoderOpusTest, OpusFlagDtxAsNonSpeech) {
+  // Create encoder with DTX enabled.
+  AudioEncoderOpusConfig config;
+  config.dtx_enabled = true;
+  config.sample_rate_hz = sample_rate_hz_;
+  constexpr int payload_type = 17;
+  const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+
+  // Open file containing speech and silence.
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
+  test::AudioLoop audio_loop;
+  // Use the file as if it were sampled at our desired input rate.
+  const size_t max_loop_length_samples =
+      sample_rate_hz_ * 10;  // Max 10 second loop.
+  const size_t input_block_size_samples =
+      10 * sample_rate_hz_ / 1000;  // 10 ms.
+  EXPECT_TRUE(audio_loop.Init(kInputFileName, max_loop_length_samples,
+                              input_block_size_samples));
+
+  // Encode.
+  AudioEncoder::EncodedInfo info;
+  rtc::Buffer encoded(500);
+  int nonspeech_frames = 0;
+  int max_nonspeech_frames = 0;
+  int dtx_frames = 0;
+  int max_dtx_frames = 0;
+  uint32_t rtp_timestamp = 0u;
+  for (size_t i = 0; i < 500; ++i) {
+    encoded.Clear();
+
+    // Every second call to the encoder will generate an Opus packet.
+    for (int j = 0; j < 2; j++) {
+      info =
+          encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
+      rtp_timestamp += input_block_size_samples;
+    }
+
+    // Bookkeeping of number of DTX frames.
+    if (info.encoded_bytes <= 2) {
+      ++dtx_frames;
+    } else {
+      if (dtx_frames > max_dtx_frames)
+        max_dtx_frames = dtx_frames;
+      dtx_frames = 0;
+    }
+
+    // Bookkeeping of number of non-speech frames.
+    if (info.speech == 0) {
+      ++nonspeech_frames;
+    } else {
+      if (nonspeech_frames > max_nonspeech_frames)
+        max_nonspeech_frames = nonspeech_frames;
+      nonspeech_frames = 0;
+    }
+  }
+
+  // Maximum number of consecutive non-speech packets should exceed 15.
+  EXPECT_GT(max_nonspeech_frames, 15);
+}
+
+TEST(AudioEncoderOpusTest, OpusDtxFilteringHighEnergyRefreshPackets) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-OpusAvoidNoisePumpingDuringDtx/Enabled/");
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/testfile16kHz", "pcm");
+  constexpr int kSampleRateHz = 16000;
+  AudioEncoderOpusConfig config;
+  config.dtx_enabled = true;
+  config.sample_rate_hz = kSampleRateHz;
+  constexpr int payload_type = 17;
+  const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+  test::AudioLoop audio_loop;
+  constexpr size_t kMaxLoopLengthSaples = kSampleRateHz * 11.6f;
+  constexpr size_t kInputBlockSizeSamples = kSampleRateHz / 100;
+  EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSaples,
+                              kInputBlockSizeSamples));
+  AudioEncoder::EncodedInfo info;
+  rtc::Buffer encoded(500);
+  // Encode the audio file and store the last part that corresponds to silence.
+  constexpr size_t kSilenceDurationSamples = kSampleRateHz * 0.2f;
+  std::array<int16_t, kSilenceDurationSamples> silence;
+  uint32_t rtp_timestamp = 0;
+  bool last_packet_dtx_frame = false;
+  bool opus_entered_dtx = false;
+  bool silence_filled = false;
+  size_t timestamp_start_silence = 0;
+  while (!silence_filled && rtp_timestamp < kMaxLoopLengthSaples) {
+    encoded.Clear();
+    // Every second call to the encoder will generate an Opus packet.
+    for (int j = 0; j < 2; j++) {
+      auto next_frame = audio_loop.GetNextBlock();
+      info = encoder->Encode(rtp_timestamp, next_frame, &encoded);
+      if (opus_entered_dtx) {
+        size_t silence_frame_start = rtp_timestamp - timestamp_start_silence;
+        silence_filled = silence_frame_start >= kSilenceDurationSamples;
+        if (!silence_filled) {
+          std::copy(next_frame.begin(), next_frame.end(),
+                    silence.begin() + silence_frame_start);
+        }
+      }
+      rtp_timestamp += kInputBlockSizeSamples;
+    }
+    EXPECT_TRUE(info.encoded_bytes > 0 || last_packet_dtx_frame);
+    last_packet_dtx_frame = info.encoded_bytes > 0 ? info.encoded_bytes <= 2
+                                                   : last_packet_dtx_frame;
+    if (info.encoded_bytes <= 2 && !opus_entered_dtx) {
+      timestamp_start_silence = rtp_timestamp;
+    }
+    opus_entered_dtx = info.encoded_bytes <= 2;
+  }
+
+  EXPECT_TRUE(silence_filled);
+  // The copied 200 ms of silence is used for creating 6 bursts that are fed to
+  // the encoder, the first three ones with a larger energy and the last three
+  // with a lower energy. This test verifies that the encoder just sends refresh
+  // DTX packets during the last bursts.
+  int number_non_empty_packets_during_increase = 0;
+  int number_non_empty_packets_during_decrease = 0;
+  for (size_t burst = 0; burst < 6; ++burst) {
+    uint32_t rtp_timestamp_start = rtp_timestamp;
+    const bool increase_noise = burst < 3;
+    const float gain = increase_noise ? 1.4f : 0.0f;
+    while (rtp_timestamp < rtp_timestamp_start + kSilenceDurationSamples) {
+      encoded.Clear();
+      // Every second call to the encoder will generate an Opus packet.
+      for (int j = 0; j < 2; j++) {
+        std::array<int16_t, kInputBlockSizeSamples> silence_frame;
+        size_t silence_frame_start = rtp_timestamp - rtp_timestamp_start;
+        std::transform(
+            silence.begin() + silence_frame_start,
+            silence.begin() + silence_frame_start + kInputBlockSizeSamples,
+            silence_frame.begin(), [gain](float s) { return gain * s; });
+        info = encoder->Encode(rtp_timestamp, silence_frame, &encoded);
+        rtp_timestamp += kInputBlockSizeSamples;
+      }
+      EXPECT_TRUE(info.encoded_bytes > 0 || last_packet_dtx_frame);
+      last_packet_dtx_frame = info.encoded_bytes > 0 ? info.encoded_bytes <= 2
+                                                     : last_packet_dtx_frame;
+      // Tracking the number of non empty packets.
+      if (increase_noise && info.encoded_bytes > 2) {
+        number_non_empty_packets_during_increase++;
+      }
+      if (!increase_noise && info.encoded_bytes > 2) {
+        number_non_empty_packets_during_decrease++;
+      }
+    }
+  }
+  // Check that the refresh DTX packets are just sent during the decrease energy
+  // region.
+  EXPECT_EQ(number_non_empty_packets_during_increase, 0);
+  EXPECT_GT(number_non_empty_packets_during_decrease, 0);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
new file mode 100644
index 0000000000..38b60c6187
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_decoder_opus.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "common_audio/include/audio_util.h"
+#include "common_audio/window_generator.h"
+#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace {
+
+constexpr size_t kNumChannels = 1u;
+constexpr int kSampleRateHz = 48000;
+constexpr size_t kMaxLoopLengthSamples = kSampleRateHz * 50;  // 50 seconds.
+constexpr size_t kInputBlockSizeSamples = 10 * kSampleRateHz / 1000;   // 10 ms
+constexpr size_t kOutputBlockSizeSamples = 20 * kSampleRateHz / 1000;  // 20 ms
+constexpr size_t kFftSize = 1024;
+constexpr size_t kNarrowbandSize = 4000 * kFftSize / kSampleRateHz;
+constexpr float kKbdAlpha = 1.5f;
+
+class PowerRatioEstimator : public LappedTransform::Callback {
+ public:
+  PowerRatioEstimator() : low_pow_(0.f), high_pow_(0.f) {
+    WindowGenerator::KaiserBesselDerived(kKbdAlpha, kFftSize, window_);
+    transform_.reset(new LappedTransform(kNumChannels, 0u,
+                                         kInputBlockSizeSamples, window_,
+                                         kFftSize, kFftSize / 2, this));
+  }
+
+  void ProcessBlock(float* data) { transform_->ProcessChunk(&data, nullptr); }
+
+  float PowerRatio() { return high_pow_ / low_pow_; }
+
+ protected:
+  void ProcessAudioBlock(const std::complex<float>* const* input,
+                         size_t num_input_channels,
+                         size_t num_freq_bins,
+                         size_t num_output_channels,
+                         std::complex<float>* const* output) override {
+    float low_pow = 0.f;
+    float high_pow = 0.f;
+    for (size_t i = 0u; i < num_input_channels; ++i) {
+      for (size_t j = 0u; j < kNarrowbandSize; ++j) {
+        float low_mag = std::abs(input[i][j]);
+        low_pow += low_mag * low_mag;
+        float high_mag = std::abs(input[i][j + kNarrowbandSize]);
+        high_pow += high_mag * high_mag;
+      }
+    }
+    low_pow_ += low_pow / (num_input_channels * kFftSize);
+    high_pow_ += high_pow / (num_input_channels * kFftSize);
+  }
+
+ private:
+  std::unique_ptr<LappedTransform> transform_;
+  float window_[kFftSize];
+  float low_pow_;
+  float high_pow_;
+};
+
+float EncodedPowerRatio(AudioEncoder* encoder,
+                        AudioDecoder* decoder,
+                        test::AudioLoop* audio_loop) {
+  // Encode and decode.
+  uint32_t rtp_timestamp = 0u;
+  constexpr size_t kBufferSize = 500;
+  rtc::Buffer encoded(kBufferSize);
+  std::vector<int16_t> decoded(kOutputBlockSizeSamples);
+  std::vector<float> decoded_float(kOutputBlockSizeSamples);
+  AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
+  PowerRatioEstimator power_ratio_estimator;
+  for (size_t i = 0; i < 1000; ++i) {
+    encoded.Clear();
+    AudioEncoder::EncodedInfo encoder_info =
+        encoder->Encode(rtp_timestamp, audio_loop->GetNextBlock(), &encoded);
+    rtp_timestamp += kInputBlockSizeSamples;
+    if (encoded.size() > 0) {
+      int decoder_info = decoder->Decode(
+          encoded.data(), encoded.size(), kSampleRateHz,
+          decoded.size() * sizeof(decoded[0]), decoded.data(), &speech_type);
+      if (decoder_info > 0) {
+        S16ToFloat(decoded.data(), decoded.size(), decoded_float.data());
+        power_ratio_estimator.ProcessBlock(decoded_float.data());
+      }
+    }
+  }
+  return power_ratio_estimator.PowerRatio();
+}
+
+}  // namespace
+
+// TODO(ivoc): Remove this test, WebRTC-AdjustOpusBandwidth is obsolete.
+TEST(BandwidthAdaptationTest, BandwidthAdaptationTest) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-AdjustOpusBandwidth/Enabled/");
+
+  constexpr float kMaxNarrowbandRatio = 0.0035f;
+  constexpr float kMinWidebandRatio = 0.01f;
+
+  // Create encoder.
+  AudioEncoderOpusConfig enc_config;
+  enc_config.bitrate_bps = absl::optional<int>(7999);
+  enc_config.num_channels = kNumChannels;
+  constexpr int payload_type = 17;
+  auto encoder = AudioEncoderOpus::MakeAudioEncoder(enc_config, payload_type);
+
+  // Create decoder.
+  AudioDecoderOpus::Config dec_config;
+  dec_config.num_channels = kNumChannels;
+  auto decoder = AudioDecoderOpus::MakeAudioDecoder(dec_config);
+
+  // Open speech file.
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
+  test::AudioLoop audio_loop;
+  EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz());
+  ASSERT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+                              kInputBlockSizeSamples));
+
+  EXPECT_LT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMaxNarrowbandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(9000);
+  EXPECT_LT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMaxNarrowbandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(9001);
+  EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMinWidebandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(8000);
+  EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMinWidebandRatio);
+
+  encoder->OnReceivedTargetAudioBitrate(12001);
+  EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
+            kMinWidebandRatio);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
new file mode 100644
index 0000000000..e8c131092c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metric.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/time_utils.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::test::GetGlobalMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+int64_t RunComplexityTest(const AudioEncoderOpusConfig& config) {
+  // Create encoder.
+  constexpr int payload_type = 17;
+  const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
+  // Open speech file.
+  const std::string kInputFileName =
+      webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
+  test::AudioLoop audio_loop;
+  constexpr int kSampleRateHz = 48000;
+  EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz());
+  constexpr size_t kMaxLoopLengthSamples =
+      kSampleRateHz * 10;  // 10 second loop.
+  constexpr size_t kInputBlockSizeSamples =
+      10 * kSampleRateHz / 1000;  // 60 ms.
+  EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
+                              kInputBlockSizeSamples));
+  // Encode.
+  const int64_t start_time_ms = rtc::TimeMillis();
+  AudioEncoder::EncodedInfo info;
+  rtc::Buffer encoded(500);
+  uint32_t rtp_timestamp = 0u;
+  for (size_t i = 0; i < 10000; ++i) {
+    encoded.Clear();
+    info = encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
+    rtp_timestamp += kInputBlockSizeSamples;
+  }
+  return rtc::TimeMillis() - start_time_ms;
+}
+
+// This test encodes an audio file using Opus twice with different bitrates
+// (~11 kbps and 15.5 kbps). The runtime for each is measured, and the ratio
+// between the two is calculated and tracked. This test explicitly sets the
+// low_rate_complexity to 9. When running on desktop platforms, this is the same
+// as the regular complexity, and the expectation is that the resulting ratio
+// should be less than 100% (since the encoder runs faster at lower bitrates,
+// given a fixed complexity setting). On the other hand, when running on
+// mobiles, the regular complexity is 5, and we expect the resulting ratio to
+// be higher, since we have explicitly asked for a higher complexity setting at
+// the lower rate.
+TEST(AudioEncoderOpusComplexityAdaptationTest, Adaptation_On) {
+  // Create config.
+  AudioEncoderOpusConfig config;
+  // The limit -- including the hysteresis window -- at which the complexity
+  // shuold be increased.
+  config.bitrate_bps = 11000 - 1;
+  config.low_rate_complexity = 9;
+  int64_t runtime_10999bps = RunComplexityTest(config);
+
+  config.bitrate_bps = 15500;
+  int64_t runtime_15500bps = RunComplexityTest(config);
+
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "opus_encoding_complexity_ratio", "adaptation_on",
+      100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+// This test is identical to the one above, but without the complexity
+// adaptation enabled (neither on desktop, nor on mobile). The expectation is
+// that the resulting ratio is less than 100% at all times.
+TEST(AudioEncoderOpusComplexityAdaptationTest, Adaptation_Off) {
+  // Create config.
+  AudioEncoderOpusConfig config;
+  // The limit -- including the hysteresis window -- at which the complexity
+  // shuold be increased (but not in this test since complexity adaptation is
+  // disabled).
+  config.bitrate_bps = 11000 - 1;
+  int64_t runtime_10999bps = RunComplexityTest(config);
+
+  config.bitrate_bps = 15500;
+  int64_t runtime_15500bps = RunComplexityTest(config);
+
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "opus_encoding_complexity_ratio", "adaptation_off",
+      100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
new file mode 100644
index 0000000000..815f26e31c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -0,0 +1,248 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+using std::get;
+using std::string;
+using std::tuple;
+using ::testing::TestWithParam;
+
+namespace webrtc {
+
+// Define coding parameter as <channels, bit_rate, filename, extension>.
+typedef tuple<size_t, int, string, string> coding_param;
+typedef struct mode mode;
+
+struct mode {
+  bool fec;
+  uint8_t target_packet_loss_rate;
+};
+
+const int kOpusBlockDurationMs = 20;
+const int kOpusSamplingKhz = 48;
+
+class OpusFecTest : public TestWithParam<coding_param> {
+ protected:
+  OpusFecTest();
+
+  void SetUp() override;
+  void TearDown() override;
+
+  virtual void EncodeABlock();
+
+  virtual void DecodeABlock(bool lost_previous, bool lost_current);
+
+  int block_duration_ms_;
+  int sampling_khz_;
+  size_t block_length_sample_;
+
+  size_t channels_;
+  int bit_rate_;
+
+  size_t data_pointer_;
+  size_t loop_length_samples_;
+  size_t max_bytes_;
+  size_t encoded_bytes_;
+
+  WebRtcOpusEncInst* opus_encoder_;
+  WebRtcOpusDecInst* opus_decoder_;
+
+  string in_filename_;
+
+  std::unique_ptr<int16_t[]> in_data_;
+  std::unique_ptr<int16_t[]> out_data_;
+  std::unique_ptr<uint8_t[]> bit_stream_;
+};
+
+void OpusFecTest::SetUp() {
+  channels_ = get<0>(GetParam());
+  bit_rate_ = get<1>(GetParam());
+  printf("Coding %zu channel signal at %d bps.\n", channels_, bit_rate_);
+
+  in_filename_ = test::ResourcePath(get<2>(GetParam()), get<3>(GetParam()));
+
+  FILE* fp = fopen(in_filename_.c_str(), "rb");
+  ASSERT_FALSE(fp == NULL);
+
+  // Obtain file size.
+  fseek(fp, 0, SEEK_END);
+  loop_length_samples_ = ftell(fp) / sizeof(int16_t);
+  rewind(fp);
+
+  // Allocate memory to contain the whole file.
+  in_data_.reset(
+      new int16_t[loop_length_samples_ + block_length_sample_ * channels_]);
+
+  // Copy the file into the buffer.
+  ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
+            loop_length_samples_);
+  fclose(fp);
+
+  // The audio will be used in a looped manner. To ease the acquisition of an
+  // audio frame that crosses the end of the excerpt, we add an extra block
+  // length of samples to the end of the array, starting over again from the
+  // beginning of the array. Audio frames cross the end of the excerpt always
+  // appear as a continuum of memory.
+  memcpy(&in_data_[loop_length_samples_], &in_data_[0],
+         block_length_sample_ * channels_ * sizeof(int16_t));
+
+  // Maximum number of bytes in output bitstream.
+  max_bytes_ = block_length_sample_ * channels_ * sizeof(int16_t);
+
+  out_data_.reset(new int16_t[2 * block_length_sample_ * channels_]);
+  bit_stream_.reset(new uint8_t[max_bytes_]);
+
+  // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
+  int app = channels_ == 1 ? 0 : 1;
+
+  // Create encoder memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000));
+  EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000));
+  // Set bitrate.
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
+}
+
+void OpusFecTest::TearDown() {
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+OpusFecTest::OpusFecTest()
+    : block_duration_ms_(kOpusBlockDurationMs),
+      sampling_khz_(kOpusSamplingKhz),
+      block_length_sample_(
+          static_cast<size_t>(block_duration_ms_ * sampling_khz_)),
+      data_pointer_(0),
+      max_bytes_(0),
+      encoded_bytes_(0),
+      opus_encoder_(NULL),
+      opus_decoder_(NULL) {}
+
+void OpusFecTest::EncodeABlock() {
+  int value =
+      WebRtcOpus_Encode(opus_encoder_, &in_data_[data_pointer_],
+                        block_length_sample_, max_bytes_, &bit_stream_[0]);
+  EXPECT_GT(value, 0);
+
+  encoded_bytes_ = static_cast<size_t>(value);
+}
+
+void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
+  int16_t audio_type;
+  int value_1 = 0, value_2 = 0;
+
+  if (lost_previous) {
+    // Decode previous frame.
+    if (!lost_current &&
+        WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_) == 1) {
+      value_1 =
+          WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0], encoded_bytes_,
+                               &out_data_[0], &audio_type);
+    } else {
+      // Call decoder PLC.
+      while (value_1 < static_cast<int>(block_length_sample_)) {
+        int ret = WebRtcOpus_Decode(opus_decoder_, NULL, 0, &out_data_[value_1],
+                                    &audio_type);
+        EXPECT_EQ(ret, sampling_khz_ * 10);  // Should return 10 ms of samples.
+        value_1 += ret;
+      }
+    }
+    EXPECT_EQ(static_cast<int>(block_length_sample_), value_1);
+  }
+
+  if (!lost_current) {
+    // Decode current frame.
+    value_2 = WebRtcOpus_Decode(opus_decoder_, &bit_stream_[0], encoded_bytes_,
+                                &out_data_[value_1 * channels_], &audio_type);
+    EXPECT_EQ(static_cast<int>(block_length_sample_), value_2);
+  }
+}
+
+TEST_P(OpusFecTest, RandomPacketLossTest) {
+  const int kDurationMs = 200000;
+  int time_now_ms, fec_frames;
+  int actual_packet_loss_rate;
+  bool lost_current, lost_previous;
+  mode mode_set[3] = {{true, 0}, {false, 0}, {true, 50}};
+
+  lost_current = false;
+  for (int i = 0; i < 3; i++) {
+    if (mode_set[i].fec) {
+      EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
+      EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(
+                       opus_encoder_, mode_set[i].target_packet_loss_rate));
+      printf("FEC is ON, target at packet loss rate %d percent.\n",
+             mode_set[i].target_packet_loss_rate);
+    } else {
+      EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
+      printf("FEC is OFF.\n");
+    }
+    // In this test, we let the target packet loss rate match the actual rate.
+    actual_packet_loss_rate = mode_set[i].target_packet_loss_rate;
+    // Run every mode a certain time.
+    time_now_ms = 0;
+    fec_frames = 0;
+    while (time_now_ms < kDurationMs) {
+      // Encode & decode.
+      EncodeABlock();
+
+      // Check if payload has FEC.
+      int fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
+
+      // If FEC is disabled or the target packet loss rate is set to 0, there
+      // should be no FEC in the bit stream.
+      if (!mode_set[i].fec || mode_set[i].target_packet_loss_rate == 0) {
+        EXPECT_EQ(fec, 0);
+      } else if (fec == 1) {
+        fec_frames++;
+      }
+
+      lost_previous = lost_current;
+      lost_current = rand() < actual_packet_loss_rate * (RAND_MAX / 100);
+      DecodeABlock(lost_previous, lost_current);
+
+      time_now_ms += block_duration_ms_;
+
+      // `data_pointer_` is incremented and wrapped across
+      // `loop_length_samples_`.
+      data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) %
+                      loop_length_samples_;
+    }
+    if (mode_set[i].fec) {
+      printf("%.2f percent frames has FEC.\n",
+             static_cast<float>(fec_frames) * block_duration_ms_ / 2000);
+    }
+  }
+}
+
+const coding_param param_set[] = {
+    std::make_tuple(1,
+                    64000,
+                    string("audio_coding/testfile32kHz"),
+                    string("pcm")),
+    std::make_tuple(1,
+                    32000,
+                    string("audio_coding/testfile32kHz"),
+                    string("pcm")),
+    std::make_tuple(2,
+                    64000,
+                    string("audio_coding/teststereo32kHz"),
+                    string("pcm"))};
+
+// 64 kbps, stereo
+INSTANTIATE_TEST_SUITE_P(AllTest, OpusFecTest, ::testing::ValuesIn(param_set));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h
new file mode 100644
index 0000000000..92c5c354a7
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_inst.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+
+#include <stddef.h>
+
+#include "rtc_base/ignore_wundef.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#include "third_party/opus/src/include/opus.h"
+#include "third_party/opus/src/include/opus_multistream.h"
+RTC_POP_IGNORING_WUNDEF()
+
+struct WebRtcOpusEncInst {
+  OpusEncoder* encoder;
+  OpusMSEncoder* multistream_encoder;
+  size_t channels;
+  int in_dtx_mode;
+  bool avoid_noise_pumping_during_dtx;
+  int sample_rate_hz;
+  float smooth_energy_non_active_frames;
+};
+
+struct WebRtcOpusDecInst {
+  OpusDecoder* decoder;
+  OpusMSDecoder* multistream_decoder;
+  int prev_decoded_samples;
+  bool plc_use_prev_decoded_samples;
+  size_t channels;
+  int in_dtx_mode;
+  int sample_rate_hz;
+};
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc
new file mode 100644
index 0000000000..64a1f59237
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.cc
@@ -0,0 +1,880 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+
+#include <cstdlib>
+#include <numeric>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+enum {
+#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
+  /* Maximum supported frame size in WebRTC is 120 ms. */
+  kWebRtcOpusMaxEncodeFrameSizeMs = 120,
+#else
+  /* Maximum supported frame size in WebRTC is 60 ms. */
+  kWebRtcOpusMaxEncodeFrameSizeMs = 60,
+#endif
+
+  /* The format allows up to 120 ms frames. Since we don't control the other
+   * side, we must allow for packets of that size. NetEq is currently limited
+   * to 60 ms on the receive side. */
+  kWebRtcOpusMaxDecodeFrameSizeMs = 120,
+
+  // Duration of audio that each call to packet loss concealment covers.
+  kWebRtcOpusPlcFrameSizeMs = 10,
+};
+
+constexpr char kPlcUsePrevDecodedSamplesFieldTrial[] =
+    "WebRTC-Audio-OpusPlcUsePrevDecodedSamples";
+
+constexpr char kAvoidNoisePumpingDuringDtxFieldTrial[] =
+    "WebRTC-Audio-OpusAvoidNoisePumpingDuringDtx";
+
+constexpr char kSetSignalVoiceWithDtxFieldTrial[] =
+    "WebRTC-Audio-OpusSetSignalVoiceWithDtx";
+
+static int FrameSizePerChannel(int frame_size_ms, int sample_rate_hz) {
+  RTC_DCHECK_GT(frame_size_ms, 0);
+  RTC_DCHECK_EQ(frame_size_ms % 10, 0);
+  RTC_DCHECK_GT(sample_rate_hz, 0);
+  RTC_DCHECK_EQ(sample_rate_hz % 1000, 0);
+  return frame_size_ms * (sample_rate_hz / 1000);
+}
+
+// Maximum sample count per channel.
+static int MaxFrameSizePerChannel(int sample_rate_hz) {
+  return FrameSizePerChannel(kWebRtcOpusMaxDecodeFrameSizeMs, sample_rate_hz);
+}
+
+// Default sample count per channel.
+static int DefaultFrameSizePerChannel(int sample_rate_hz) {
+  return FrameSizePerChannel(20, sample_rate_hz);
+}
+
+// Returns true if the `encoded` payload corresponds to a refresh DTX packet
+// whose energy is larger than the expected for non activity packets.
+static bool WebRtcOpus_IsHighEnergyRefreshDtxPacket(
+    OpusEncInst* inst,
+    rtc::ArrayView<const int16_t> frame,
+    rtc::ArrayView<const uint8_t> encoded) {
+  if (encoded.size() <= 2) {
+    return false;
+  }
+  int number_frames =
+      frame.size() / DefaultFrameSizePerChannel(inst->sample_rate_hz);
+  if (number_frames > 0 &&
+      WebRtcOpus_PacketHasVoiceActivity(encoded.data(), encoded.size()) == 0) {
+    const float average_frame_energy =
+        std::accumulate(frame.begin(), frame.end(), 0.0f,
+                        [](float a, int32_t b) { return a + b * b; }) /
+        number_frames;
+    if (WebRtcOpus_GetInDtx(inst) == 1 &&
+        average_frame_energy >= inst->smooth_energy_non_active_frames * 0.5f) {
+      // This is a refresh DTX packet as the encoder is in DTX and has
+      // produced a payload > 2 bytes. This refresh packet has a higher energy
+      // than the smooth energy of non activity frames (with a 3 dB negative
+      // margin) and, therefore, it is flagged as a high energy refresh DTX
+      // packet.
+      return true;
+    }
+    // The average energy is tracked in a similar way as the modeling of the
+    // comfort noise in the Silk decoder in Opus
+    // (third_party/opus/src/silk/CNG.c).
+    if (average_frame_energy < inst->smooth_energy_non_active_frames * 0.5f) {
+      inst->smooth_energy_non_active_frames = average_frame_energy;
+    } else {
+      inst->smooth_energy_non_active_frames +=
+          (average_frame_energy - inst->smooth_energy_non_active_frames) *
+          0.25f;
+    }
+  }
+  return false;
+}
+
+int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
+                                 size_t channels,
+                                 int32_t application,
+                                 int sample_rate_hz) {
+  int opus_app;
+  if (!inst)
+    return -1;
+
+  switch (application) {
+    case 0:
+      opus_app = OPUS_APPLICATION_VOIP;
+      break;
+    case 1:
+      opus_app = OPUS_APPLICATION_AUDIO;
+      break;
+    default:
+      return -1;
+  }
+
+  OpusEncInst* state =
+      reinterpret_cast<OpusEncInst*>(calloc(1, sizeof(OpusEncInst)));
+  RTC_DCHECK(state);
+
+  int error;
+  state->encoder = opus_encoder_create(
+      sample_rate_hz, static_cast<int>(channels), opus_app, &error);
+
+  if (error != OPUS_OK || (!state->encoder && !state->multistream_encoder)) {
+    WebRtcOpus_EncoderFree(state);
+    return -1;
+  }
+
+  state->in_dtx_mode = 0;
+  state->channels = channels;
+  state->sample_rate_hz = sample_rate_hz;
+  state->smooth_energy_non_active_frames = 0.0f;
+  state->avoid_noise_pumping_during_dtx =
+      webrtc::field_trial::IsEnabled(kAvoidNoisePumpingDuringDtxFieldTrial);
+
+  *inst = state;
+  return 0;
+}
+
+int16_t WebRtcOpus_MultistreamEncoderCreate(
+    OpusEncInst** inst,
+    size_t channels,
+    int32_t application,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping) {
+  int opus_app;
+  if (!inst)
+    return -1;
+
+  switch (application) {
+    case 0:
+      opus_app = OPUS_APPLICATION_VOIP;
+      break;
+    case 1:
+      opus_app = OPUS_APPLICATION_AUDIO;
+      break;
+    default:
+      return -1;
+  }
+
+  OpusEncInst* state =
+      reinterpret_cast<OpusEncInst*>(calloc(1, sizeof(OpusEncInst)));
+  RTC_DCHECK(state);
+
+  int error;
+  const int sample_rate_hz = 48000;
+  state->multistream_encoder = opus_multistream_encoder_create(
+      sample_rate_hz, channels, streams, coupled_streams, channel_mapping,
+      opus_app, &error);
+
+  if (error != OPUS_OK || (!state->encoder && !state->multistream_encoder)) {
+    WebRtcOpus_EncoderFree(state);
+    return -1;
+  }
+
+  state->in_dtx_mode = 0;
+  state->channels = channels;
+  state->sample_rate_hz = sample_rate_hz;
+  state->smooth_energy_non_active_frames = 0.0f;
+  state->avoid_noise_pumping_during_dtx = false;
+
+  *inst = state;
+  return 0;
+}
+
+int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst) {
+  if (inst) {
+    if (inst->encoder) {
+      opus_encoder_destroy(inst->encoder);
+    } else {
+      opus_multistream_encoder_destroy(inst->multistream_encoder);
+    }
+    free(inst);
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+int WebRtcOpus_Encode(OpusEncInst* inst,
+                      const int16_t* audio_in,
+                      size_t samples,
+                      size_t length_encoded_buffer,
+                      uint8_t* encoded) {
+  int res;
+
+  if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
+    return -1;
+  }
+
+  if (inst->encoder) {
+    res = opus_encode(inst->encoder, (const opus_int16*)audio_in,
+                      static_cast<int>(samples), encoded,
+                      static_cast<opus_int32>(length_encoded_buffer));
+  } else {
+    res = opus_multistream_encode(
+        inst->multistream_encoder, (const opus_int16*)audio_in,
+        static_cast<int>(samples), encoded,
+        static_cast<opus_int32>(length_encoded_buffer));
+  }
+
+  if (res <= 0) {
+    return -1;
+  }
+
+  if (res <= 2) {
+    // Indicates DTX since the packet has nothing but a header. In principle,
+    // there is no need to send this packet. However, we do transmit the first
+    // occurrence to let the decoder know that the encoder enters DTX mode.
+    if (inst->in_dtx_mode) {
+      return 0;
+    } else {
+      inst->in_dtx_mode = 1;
+      return res;
+    }
+  }
+
+  if (inst->avoid_noise_pumping_during_dtx && WebRtcOpus_GetUseDtx(inst) == 1 &&
+      WebRtcOpus_IsHighEnergyRefreshDtxPacket(
+          inst, rtc::MakeArrayView(audio_in, samples),
+          rtc::MakeArrayView(encoded, res))) {
+    // This packet is a high energy refresh DTX packet. For avoiding an increase
+    // of the energy in the DTX region at the decoder, this packet is
+    // substituted by a TOC byte with one empty frame.
+    // The number of frames described in the TOC byte
+    // (https://tools.ietf.org/html/rfc6716#section-3.1) are overwritten to
+    // always indicate one frame (last two bits equal to 0).
+    encoded[0] = encoded[0] & 0b11111100;
+    inst->in_dtx_mode = 1;
+    // The payload is just the TOC byte and has 1 byte as length.
+    return 1;
+  }
+  inst->in_dtx_mode = 0;
+  return res;
+}
+
+#define ENCODER_CTL(inst, vargs)                \
+  (inst->encoder                                \
+       ? opus_encoder_ctl(inst->encoder, vargs) \
+       : opus_multistream_encoder_ctl(inst->multistream_encoder, vargs))
+
+int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_BITRATE(rate));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_PACKET_LOSS_PERC(loss_rate));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz) {
+  opus_int32 set_bandwidth;
+
+  if (!inst)
+    return -1;
+
+  if (frequency_hz <= 8000) {
+    set_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+  } else if (frequency_hz <= 12000) {
+    set_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+  } else if (frequency_hz <= 16000) {
+    set_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+  } else if (frequency_hz <= 24000) {
+    set_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
+  } else {
+    set_bandwidth = OPUS_BANDWIDTH_FULLBAND;
+  }
+  return ENCODER_CTL(inst, OPUS_SET_MAX_BANDWIDTH(set_bandwidth));
+}
+
+int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
+                                      int32_t* result_hz) {
+  if (inst->encoder) {
+    if (opus_encoder_ctl(inst->encoder, OPUS_GET_MAX_BANDWIDTH(result_hz)) ==
+        OPUS_OK) {
+      return 0;
+    }
+    return -1;
+  }
+
+  opus_int32 max_bandwidth;
+  int s;
+  int ret;
+
+  max_bandwidth = 0;
+  ret = OPUS_OK;
+  s = 0;
+  while (ret == OPUS_OK) {
+    OpusEncoder* enc;
+    opus_int32 bandwidth;
+
+    ret = ENCODER_CTL(inst, OPUS_MULTISTREAM_GET_ENCODER_STATE(s, &enc));
+    if (ret == OPUS_BAD_ARG)
+      break;
+    if (ret != OPUS_OK)
+      return -1;
+    if (opus_encoder_ctl(enc, OPUS_GET_MAX_BANDWIDTH(&bandwidth)) != OPUS_OK)
+      return -1;
+
+    if (max_bandwidth != 0 && max_bandwidth != bandwidth)
+      return -1;
+
+    max_bandwidth = bandwidth;
+    s++;
+  }
+  *result_hz = max_bandwidth;
+  return 0;
+}
+
+int16_t WebRtcOpus_EnableFec(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_INBAND_FEC(1));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_DisableFec(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_INBAND_FEC(0));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst) {
+  if (inst) {
+    if (webrtc::field_trial::IsEnabled(kSetSignalVoiceWithDtxFieldTrial)) {
+      int ret = ENCODER_CTL(inst, OPUS_SET_SIGNAL(OPUS_SIGNAL_VOICE));
+      if (ret != OPUS_OK) {
+        return ret;
+      }
+    }
+    return ENCODER_CTL(inst, OPUS_SET_DTX(1));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst) {
+  if (inst) {
+    if (webrtc::field_trial::IsEnabled(kSetSignalVoiceWithDtxFieldTrial)) {
+      int ret = ENCODER_CTL(inst, OPUS_SET_SIGNAL(OPUS_AUTO));
+      if (ret != OPUS_OK) {
+        return ret;
+      }
+    }
+    return ENCODER_CTL(inst, OPUS_SET_DTX(0));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_GetUseDtx(OpusEncInst* inst) {
+  if (inst) {
+    opus_int32 use_dtx;
+    if (ENCODER_CTL(inst, OPUS_GET_DTX(&use_dtx)) == 0) {
+      return use_dtx;
+    }
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_EnableCbr(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_VBR(0));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_DisableCbr(OpusEncInst* inst) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_VBR(1));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_COMPLEXITY(complexity));
+  } else {
+    return -1;
+  }
+}
+
+int32_t WebRtcOpus_GetBandwidth(OpusEncInst* inst) {
+  if (!inst) {
+    return -1;
+  }
+  int32_t bandwidth;
+  if (ENCODER_CTL(inst, OPUS_GET_BANDWIDTH(&bandwidth)) == 0) {
+    return bandwidth;
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth) {
+  if (inst) {
+    return ENCODER_CTL(inst, OPUS_SET_BANDWIDTH(bandwidth));
+  } else {
+    return -1;
+  }
+}
+
+int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels) {
+  if (!inst)
+    return -1;
+  if (num_channels == 0) {
+    return ENCODER_CTL(inst, OPUS_SET_FORCE_CHANNELS(OPUS_AUTO));
+  } else if (num_channels == 1 || num_channels == 2) {
+    return ENCODER_CTL(inst, OPUS_SET_FORCE_CHANNELS(num_channels));
+  } else {
+    return -1;
+  }
+}
+
+int32_t WebRtcOpus_GetInDtx(OpusEncInst* inst) {
+  if (!inst) {
+    return -1;
+  }
+#ifdef OPUS_GET_IN_DTX
+  int32_t in_dtx;
+  if (ENCODER_CTL(inst, OPUS_GET_IN_DTX(&in_dtx)) == 0) {
+    return in_dtx;
+  }
+#endif
+  return -1;
+}
+
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
+                                 size_t channels,
+                                 int sample_rate_hz) {
+  int error;
+  OpusDecInst* state;
+
+  if (inst != NULL) {
+    // Create Opus decoder state.
+    state = reinterpret_cast<OpusDecInst*>(calloc(1, sizeof(OpusDecInst)));
+    if (state == NULL) {
+      return -1;
+    }
+
+    state->decoder =
+        opus_decoder_create(sample_rate_hz, static_cast<int>(channels), &error);
+    if (error == OPUS_OK && state->decoder) {
+      // Creation of memory all ok.
+      state->channels = channels;
+      state->sample_rate_hz = sample_rate_hz;
+      state->plc_use_prev_decoded_samples =
+          webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
+      if (state->plc_use_prev_decoded_samples) {
+        state->prev_decoded_samples =
+            DefaultFrameSizePerChannel(state->sample_rate_hz);
+      }
+      state->in_dtx_mode = 0;
+      *inst = state;
+      return 0;
+    }
+
+    // If memory allocation was unsuccessful, free the entire state.
+    if (state->decoder) {
+      opus_decoder_destroy(state->decoder);
+    }
+    free(state);
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_MultistreamDecoderCreate(
+    OpusDecInst** inst,
+    size_t channels,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping) {
+  int error;
+  OpusDecInst* state;
+
+  if (inst != NULL) {
+    // Create Opus decoder state.
+    state = reinterpret_cast<OpusDecInst*>(calloc(1, sizeof(OpusDecInst)));
+    if (state == NULL) {
+      return -1;
+    }
+
+    // Create new memory, always at 48000 Hz.
+    state->multistream_decoder = opus_multistream_decoder_create(
+        48000, channels, streams, coupled_streams, channel_mapping, &error);
+
+    if (error == OPUS_OK && state->multistream_decoder) {
+      // Creation of memory all ok.
+      state->channels = channels;
+      state->sample_rate_hz = 48000;
+      state->plc_use_prev_decoded_samples =
+          webrtc::field_trial::IsEnabled(kPlcUsePrevDecodedSamplesFieldTrial);
+      if (state->plc_use_prev_decoded_samples) {
+        state->prev_decoded_samples =
+            DefaultFrameSizePerChannel(state->sample_rate_hz);
+      }
+      state->in_dtx_mode = 0;
+      *inst = state;
+      return 0;
+    }
+
+    // If memory allocation was unsuccessful, free the entire state.
+    opus_multistream_decoder_destroy(state->multistream_decoder);
+    free(state);
+  }
+  return -1;
+}
+
+int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst) {
+  if (inst) {
+    if (inst->decoder) {
+      opus_decoder_destroy(inst->decoder);
+    } else if (inst->multistream_decoder) {
+      opus_multistream_decoder_destroy(inst->multistream_decoder);
+    }
+    free(inst);
+    return 0;
+  } else {
+    return -1;
+  }
+}
+
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst) {
+  return inst->channels;
+}
+
+void WebRtcOpus_DecoderInit(OpusDecInst* inst) {
+  if (inst->decoder) {
+    opus_decoder_ctl(inst->decoder, OPUS_RESET_STATE);
+  } else {
+    opus_multistream_decoder_ctl(inst->multistream_decoder, OPUS_RESET_STATE);
+  }
+  inst->in_dtx_mode = 0;
+}
+
+/* For decoder to determine if it is to output speech or comfort noise. */
+static int16_t DetermineAudioType(OpusDecInst* inst, size_t encoded_bytes) {
+  // Audio type becomes comfort noise if `encoded_byte` is 1 and keeps
+  // to be so if the following `encoded_byte` are 0 or 1.
+  if (encoded_bytes == 0 && inst->in_dtx_mode) {
+    return 2;  // Comfort noise.
+  } else if (encoded_bytes == 1 || encoded_bytes == 2) {
+    // TODO(henrik.lundin): There is a slight risk that a 2-byte payload is in
+    // fact a 1-byte TOC with a 1-byte payload. That will be erroneously
+    // interpreted as comfort noise output, but such a payload is probably
+    // faulty anyway.
+
+    // TODO(webrtc:10218): This is wrong for multistream opus. Then are several
+    // single-stream packets glued together with some packet size bytes in
+    // between. See https://tools.ietf.org/html/rfc6716#appendix-B
+    inst->in_dtx_mode = 1;
+    return 2;  // Comfort noise.
+  } else {
+    inst->in_dtx_mode = 0;
+    return 0;  // Speech.
+  }
+}
+
+/* `frame_size` is set to maximum Opus frame size in the normal case, and
+ * is set to the number of samples needed for PLC in case of losses.
+ * It is up to the caller to make sure the value is correct. */
+static int DecodeNative(OpusDecInst* inst,
+                        const uint8_t* encoded,
+                        size_t encoded_bytes,
+                        int frame_size,
+                        int16_t* decoded,
+                        int16_t* audio_type,
+                        int decode_fec) {
+  int res = -1;
+  if (inst->decoder) {
+    res = opus_decode(
+        inst->decoder, encoded, static_cast<opus_int32>(encoded_bytes),
+        reinterpret_cast<opus_int16*>(decoded), frame_size, decode_fec);
+  } else {
+    res = opus_multistream_decode(inst->multistream_decoder, encoded,
+                                  static_cast<opus_int32>(encoded_bytes),
+                                  reinterpret_cast<opus_int16*>(decoded),
+                                  frame_size, decode_fec);
+  }
+
+  if (res <= 0)
+    return -1;
+
+  *audio_type = DetermineAudioType(inst, encoded_bytes);
+
+  return res;
+}
+
+static int DecodePlc(OpusDecInst* inst, int16_t* decoded) {
+  int16_t audio_type = 0;
+  int decoded_samples;
+  int plc_samples =
+      FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
+
+  if (inst->plc_use_prev_decoded_samples) {
+    /* The number of samples we ask for is `number_of_lost_frames` times
+     * `prev_decoded_samples_`. Limit the number of samples to maximum
+     * `MaxFrameSizePerChannel()`. */
+    plc_samples = inst->prev_decoded_samples;
+    const int max_samples_per_channel =
+        MaxFrameSizePerChannel(inst->sample_rate_hz);
+    plc_samples = plc_samples <= max_samples_per_channel
+                      ? plc_samples
+                      : max_samples_per_channel;
+  }
+  decoded_samples =
+      DecodeNative(inst, NULL, 0, plc_samples, decoded, &audio_type, 0);
+  if (decoded_samples < 0) {
+    return -1;
+  }
+
+  return decoded_samples;
+}
+
+int WebRtcOpus_Decode(OpusDecInst* inst,
+                      const uint8_t* encoded,
+                      size_t encoded_bytes,
+                      int16_t* decoded,
+                      int16_t* audio_type) {
+  int decoded_samples;
+
+  if (encoded_bytes == 0) {
+    *audio_type = DetermineAudioType(inst, encoded_bytes);
+    decoded_samples = DecodePlc(inst, decoded);
+  } else {
+    decoded_samples = DecodeNative(inst, encoded, encoded_bytes,
+                                   MaxFrameSizePerChannel(inst->sample_rate_hz),
+                                   decoded, audio_type, 0);
+  }
+  if (decoded_samples < 0) {
+    return -1;
+  }
+
+  if (inst->plc_use_prev_decoded_samples) {
+    /* Update decoded sample memory, to be used by the PLC in case of losses. */
+    inst->prev_decoded_samples = decoded_samples;
+  }
+
+  return decoded_samples;
+}
+
+int WebRtcOpus_DecodeFec(OpusDecInst* inst,
+                         const uint8_t* encoded,
+                         size_t encoded_bytes,
+                         int16_t* decoded,
+                         int16_t* audio_type) {
+  int decoded_samples;
+  int fec_samples;
+
+  if (WebRtcOpus_PacketHasFec(encoded, encoded_bytes) != 1) {
+    return 0;
+  }
+
+  fec_samples =
+      opus_packet_get_samples_per_frame(encoded, inst->sample_rate_hz);
+
+  decoded_samples = DecodeNative(inst, encoded, encoded_bytes, fec_samples,
+                                 decoded, audio_type, 1);
+  if (decoded_samples < 0) {
+    return -1;
+  }
+
+  return decoded_samples;
+}
+
+int WebRtcOpus_DurationEst(OpusDecInst* inst,
+                           const uint8_t* payload,
+                           size_t payload_length_bytes) {
+  if (payload_length_bytes == 0) {
+    // WebRtcOpus_Decode calls PLC when payload length is zero. So we return
+    // PLC duration correspondingly.
+    return WebRtcOpus_PlcDuration(inst);
+  }
+
+  int frames, samples;
+  frames = opus_packet_get_nb_frames(
+      payload, static_cast<opus_int32>(payload_length_bytes));
+  if (frames < 0) {
+    /* Invalid payload data. */
+    return 0;
+  }
+  samples =
+      frames * opus_packet_get_samples_per_frame(payload, inst->sample_rate_hz);
+  if (samples > 120 * inst->sample_rate_hz / 1000) {
+    // More than 120 ms' worth of samples.
+    return 0;
+  }
+  return samples;
+}
+
+int WebRtcOpus_PlcDuration(OpusDecInst* inst) {
+  if (inst->plc_use_prev_decoded_samples) {
+    /* The number of samples we ask for is `number_of_lost_frames` times
+     * `prev_decoded_samples_`. Limit the number of samples to maximum
+     * `MaxFrameSizePerChannel()`. */
+    const int plc_samples = inst->prev_decoded_samples;
+    const int max_samples_per_channel =
+        MaxFrameSizePerChannel(inst->sample_rate_hz);
+    return plc_samples <= max_samples_per_channel ? plc_samples
+                                                  : max_samples_per_channel;
+  }
+  return FrameSizePerChannel(kWebRtcOpusPlcFrameSizeMs, inst->sample_rate_hz);
+}
+
+int WebRtcOpus_FecDurationEst(const uint8_t* payload,
+                              size_t payload_length_bytes,
+                              int sample_rate_hz) {
+  if (WebRtcOpus_PacketHasFec(payload, payload_length_bytes) != 1) {
+    return 0;
+  }
+  const int samples =
+      opus_packet_get_samples_per_frame(payload, sample_rate_hz);
+  const int samples_per_ms = sample_rate_hz / 1000;
+  if (samples < 10 * samples_per_ms || samples > 120 * samples_per_ms) {
+    /* Invalid payload duration. */
+    return 0;
+  }
+  return samples;
+}
+
+int WebRtcOpus_NumSilkFrames(const uint8_t* payload) {
+  // For computing the payload length in ms, the sample rate is not important
+  // since it cancels out. We use 48 kHz, but any valid sample rate would work.
+  int payload_length_ms =
+      opus_packet_get_samples_per_frame(payload, 48000) / 48;
+  if (payload_length_ms < 10)
+    payload_length_ms = 10;
+
+  int silk_frames;
+  switch (payload_length_ms) {
+    case 10:
+    case 20:
+      silk_frames = 1;
+      break;
+    case 40:
+      silk_frames = 2;
+      break;
+    case 60:
+      silk_frames = 3;
+      break;
+    default:
+      return 0;  // It is actually even an invalid packet.
+  }
+  return silk_frames;
+}
+
+// This method is based on Definition of the Opus Audio Codec
+// (https://tools.ietf.org/html/rfc6716). Basically, this method is based on
+// parsing the LP layer of an Opus packet, particularly the LBRR flag.
+int WebRtcOpus_PacketHasFec(const uint8_t* payload,
+                            size_t payload_length_bytes) {
+  if (payload == NULL || payload_length_bytes == 0)
+    return 0;
+
+  // In CELT_ONLY mode, packets should not have FEC.
+  if (payload[0] & 0x80)
+    return 0;
+
+  int silk_frames = WebRtcOpus_NumSilkFrames(payload);
+  if (silk_frames == 0)
+    return 0;  // Not valid.
+
+  const int channels = opus_packet_get_nb_channels(payload);
+  RTC_DCHECK(channels == 1 || channels == 2);
+
+  // Max number of frames in an Opus packet is 48.
+  opus_int16 frame_sizes[48];
+  const unsigned char* frame_data[48];
+
+  // Parse packet to get the frames. But we only care about the first frame,
+  // since we can only decode the FEC from the first one.
+  if (opus_packet_parse(payload, static_cast<opus_int32>(payload_length_bytes),
+                        NULL, frame_data, frame_sizes, NULL) < 0) {
+    return 0;
+  }
+
+  if (frame_sizes[0] < 1) {
+    return 0;
+  }
+
+  // A frame starts with the LP layer. The LP layer begins with two to eight
+  // header bits.These consist of one VAD bit per SILK frame (up to 3),
+  // followed by a single flag indicating the presence of LBRR frames.
+  // For a stereo packet, these first flags correspond to the mid channel, and
+  // a second set of flags is included for the side channel. Because these are
+  // the first symbols decoded by the range coder and because they are coded
+  // as binary values with uniform probability, they can be extracted directly
+  // from the most significant bits of the first byte of compressed data.
+  for (int n = 0; n < channels; n++) {
+    // The LBRR bit for channel 1 is on the (`silk_frames` + 1)-th bit, and
+    // that of channel 2 is on the |(`silk_frames` + 1) * 2 + 1|-th bit.
+    if (frame_data[0][0] & (0x80 >> ((n + 1) * (silk_frames + 1) - 1)))
+      return 1;
+  }
+
+  return 0;
+}
+
+int WebRtcOpus_PacketHasVoiceActivity(const uint8_t* payload,
+                                      size_t payload_length_bytes) {
+  if (payload == NULL || payload_length_bytes == 0)
+    return 0;
+
+  // In CELT_ONLY mode we can not determine whether there is VAD.
+  if (payload[0] & 0x80)
+    return -1;
+
+  int silk_frames = WebRtcOpus_NumSilkFrames(payload);
+  if (silk_frames == 0)
+    return -1;
+
+  const int channels = opus_packet_get_nb_channels(payload);
+  RTC_DCHECK(channels == 1 || channels == 2);
+
+  // Max number of frames in an Opus packet is 48.
+  opus_int16 frame_sizes[48];
+  const unsigned char* frame_data[48];
+
+  // Parse packet to get the frames.
+  int frames =
+      opus_packet_parse(payload, static_cast<opus_int32>(payload_length_bytes),
+                        NULL, frame_data, frame_sizes, NULL);
+  if (frames < 0)
+    return -1;
+
+  // Iterate over all Opus frames which may contain multiple SILK frames.
+  for (int frame = 0; frame < frames; frame++) {
+    if (frame_sizes[frame] < 1) {
+      continue;
+    }
+    if (frame_data[frame][0] >> (8 - silk_frames))
+      return 1;
+    if (channels == 2 &&
+        (frame_data[frame][0] << (silk_frames + 1)) >> (8 - silk_frames))
+      return 1;
+  }
+
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.h
new file mode 100644
index 0000000000..89159ce1c0
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_interface.h
@@ -0,0 +1,547 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+// Opaque wrapper types for the codec state.
+typedef struct WebRtcOpusEncInst OpusEncInst;
+typedef struct WebRtcOpusDecInst OpusDecInst;
+
+/****************************************************************************
+ * WebRtcOpus_EncoderCreate(...)
+ *
+ * This function creates an Opus encoder that encodes mono or stereo.
+ *
+ * Input:
+ *      - channels           : number of channels; 1 or 2.
+ *      - application        : 0 - VOIP applications.
+ *                                 Favor speech intelligibility.
+ *                             1 - Audio applications.
+ *                                 Favor faithfulness to the original input.
+ *      - sample_rate_hz     : sample rate of input audio
+ *
+ * Output:
+ *      - inst               : a pointer to Encoder context that is created
+ *                             if success.
+ *
+ * Return value              : 0 - Success
+ *                            -1 - Error
+ */
+int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst,
+                                 size_t channels,
+                                 int32_t application,
+                                 int sample_rate_hz);
+
+/****************************************************************************
+ * WebRtcOpus_MultistreamEncoderCreate(...)
+ *
+ * This function creates an Opus encoder with any supported channel count.
+ *
+ * Input:
+ *      - channels           : number of channels in the input of the encoder.
+ *      - application        : 0 - VOIP applications.
+ *                                 Favor speech intelligibility.
+ *                             1 - Audio applications.
+ *                                 Favor faithfulness to the original input.
+ *      - streams            : number of streams, as described in RFC 7845.
+ *      - coupled_streams    : number of coupled streams, as described in
+ *                             RFC 7845.
+ *      - channel_mapping    : the channel mapping; pointer to array of
+ *                             `channel` bytes, as described in RFC 7845.
+ *
+ * Output:
+ *      - inst               : a pointer to Encoder context that is created
+ *                             if success.
+ *
+ * Return value              : 0 - Success
+ *                            -1 - Error
+ */
+int16_t WebRtcOpus_MultistreamEncoderCreate(
+    OpusEncInst** inst,
+    size_t channels,
+    int32_t application,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping);
+
+int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_Encode(...)
+ *
+ * This function encodes audio as a series of Opus frames and inserts
+ * it into a packet. Input buffer can be any length.
+ *
+ * Input:
+ *      - inst                  : Encoder context
+ *      - audio_in              : Input speech data buffer
+ *      - samples               : Samples per channel in audio_in
+ *      - length_encoded_buffer : Output buffer size
+ *
+ * Output:
+ *      - encoded               : Output compressed data buffer
+ *
+ * Return value                 : >=0 - Length (in bytes) of coded data
+ *                                -1 - Error
+ */
+int WebRtcOpus_Encode(OpusEncInst* inst,
+                      const int16_t* audio_in,
+                      size_t samples,
+                      size_t length_encoded_buffer,
+                      uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcOpus_SetBitRate(...)
+ *
+ * This function adjusts the target bitrate of the encoder.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - rate               : New target bitrate
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate);
+
+/****************************************************************************
+ * WebRtcOpus_SetPacketLossRate(...)
+ *
+ * This function configures the encoder's expected packet loss percentage.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - loss_rate          : loss percentage in the range 0-100, inclusive.
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetPacketLossRate(OpusEncInst* inst, int32_t loss_rate);
+
+/****************************************************************************
+ * WebRtcOpus_SetMaxPlaybackRate(...)
+ *
+ * Configures the maximum playback rate for encoding. Due to hardware
+ * limitations, the receiver may render audio up to a playback rate. Opus
+ * encoder can use this information to optimize for network usage and encoding
+ * complexity. This will affect the audio bandwidth in the coded audio. However,
+ * the input/output sample rate is not affected.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - frequency_hz       : Maximum playback rate in Hz.
+ *                             This parameter can take any value. The relation
+ *                             between the value and the Opus internal mode is
+ *                             as following:
+ *                             frequency_hz <= 8000           narrow band
+ *                             8000 < frequency_hz <= 12000   medium band
+ *                             12000 < frequency_hz <= 16000  wide band
+ *                             16000 < frequency_hz <= 24000  super wide band
+ *                             frequency_hz > 24000           full band
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetMaxPlaybackRate(OpusEncInst* inst, int32_t frequency_hz);
+
+/****************************************************************************
+ * WebRtcOpus_GetMaxPlaybackRate(...)
+ *
+ * Queries the maximum playback rate for encoding. If different single-stream
+ * encoders have different maximum playback rates, this function fails.
+ *
+ * Input:
+ *      - inst               : Encoder context.
+ * Output:
+ *      - result_hz          : The maximum playback rate in Hz.
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_GetMaxPlaybackRate(OpusEncInst* const inst,
+                                      int32_t* result_hz);
+
+/* TODO(minyue): Check whether an API to check the FEC and the packet loss rate
+ * is needed. It might not be very useful since there are not many use cases and
+ * the caller can always maintain the states. */
+
+/****************************************************************************
+ * WebRtcOpus_EnableFec()
+ *
+ * This function enables FEC for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_EnableFec(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DisableFec()
+ *
+ * This function disables FEC for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_DisableFec(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_EnableDtx()
+ *
+ * This function enables Opus internal DTX for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_EnableDtx(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DisableDtx()
+ *
+ * This function disables Opus internal DTX for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_DisableDtx(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_GetUseDtx()
+ *
+ * This function gets the DTX configuration used for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Encoder does not use DTX.
+ *                              1 - Encoder uses DTX.
+ *                             -1 - Error.
+ */
+int16_t WebRtcOpus_GetUseDtx(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_EnableCbr()
+ *
+ * This function enables CBR for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_EnableCbr(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DisableCbr()
+ *
+ * This function disables CBR for encoding.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_DisableCbr(OpusEncInst* inst);
+
+/*
+ * WebRtcOpus_SetComplexity(...)
+ *
+ * This function adjusts the computational complexity. The effect is the same as
+ * calling the complexity setting of Opus as an Opus encoder related CTL.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - complexity         : New target complexity (0-10, inclusive)
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetComplexity(OpusEncInst* inst, int32_t complexity);
+
+/*
+ * WebRtcOpus_GetBandwidth(...)
+ *
+ * This function returns the current bandwidth.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *
+ * Return value              :  Bandwidth - Success
+ *                             -1 - Error
+ */
+int32_t WebRtcOpus_GetBandwidth(OpusEncInst* inst);
+
+/*
+ * WebRtcOpus_SetBandwidth(...)
+ *
+ * By default Opus decides which bandwidth to encode the signal in depending on
+ * the the bitrate. This function overrules the previous setting and forces the
+ * encoder to encode in narrowband/wideband/fullband/etc.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - bandwidth          : New target bandwidth. Valid values are:
+ *                             OPUS_BANDWIDTH_NARROWBAND
+ *                             OPUS_BANDWIDTH_MEDIUMBAND
+ *                             OPUS_BANDWIDTH_WIDEBAND
+ *                             OPUS_BANDWIDTH_SUPERWIDEBAND
+ *                             OPUS_BANDWIDTH_FULLBAND
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetBandwidth(OpusEncInst* inst, int32_t bandwidth);
+
+/*
+ * WebRtcOpus_GetInDtx(...)
+ *
+ * Gets the DTX state of the encoder.
+ *
+ * Input:
+ *      - inst   : Encoder context
+ *
+ * Return value  : -1 - Error.
+ *                 1  - Last encoded frame was comfort noise update during DTX.
+ *                 0  - Last encoded frame was encoded with encoder not in DTX.
+ */
+int32_t WebRtcOpus_GetInDtx(OpusEncInst* inst);
+
+/*
+ * WebRtcOpus_SetForceChannels(...)
+ *
+ * If the encoder is initialized as a stereo encoder, Opus will by default
+ * decide whether to encode in mono or stereo based on the bitrate. This
+ * function overrules the previous setting, and forces the encoder to encode
+ * in auto/mono/stereo.
+ *
+ * If the Encoder is initialized as a mono encoder, and one tries to force
+ * stereo, the function will return an error.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - num_channels       : 0 - Not forced
+ *                             1 - Mono
+ *                             2 - Stereo
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetForceChannels(OpusEncInst* inst, size_t num_channels);
+
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst,
+                                 size_t channels,
+                                 int sample_rate_hz);
+
+/****************************************************************************
+ * WebRtcOpus_MultistreamDecoderCreate(...)
+ *
+ * This function creates an Opus decoder with any supported channel count.
+ *
+ * Input:
+ *      - channels           : number of output channels that the decoder
+ *                             will produce.
+ *      - streams            : number of encoded streams, as described in
+ *                             RFC 7845.
+ *      - coupled_streams    : number of coupled streams, as described in
+ *                             RFC 7845.
+ *      - channel_mapping    : the channel mapping; pointer to array of
+ *                             `channel` bytes, as described in RFC 7845.
+ *
+ * Output:
+ *      - inst               : a pointer to a Decoder context that is created
+ *                             if success.
+ *
+ * Return value              : 0 - Success
+ *                            -1 - Error
+ */
+int16_t WebRtcOpus_MultistreamDecoderCreate(
+    OpusDecInst** inst,
+    size_t channels,
+    size_t streams,
+    size_t coupled_streams,
+    const unsigned char* channel_mapping);
+
+int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DecoderChannels(...)
+ *
+ * This function returns the number of channels created for Opus decoder.
+ */
+size_t WebRtcOpus_DecoderChannels(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DecoderInit(...)
+ *
+ * This function resets state of the decoder.
+ *
+ * Input:
+ *      - inst               : Decoder context
+ */
+void WebRtcOpus_DecoderInit(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_Decode(...)
+ *
+ * This function decodes an Opus packet into one or more audio frames at the
+ * ACM interface's sampling rate (32 kHz).
+ *
+ * Input:
+ *      - inst               : Decoder context
+ *      - encoded            : Encoded data
+ *      - encoded_bytes      : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - audio_type         : 1 normal, 2 CNG (for Opus it should
+ *                             always return 1 since we're not using Opus's
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples per channel in decoded vector
+ *                             -1 - Error
+ */
+int WebRtcOpus_Decode(OpusDecInst* inst,
+                      const uint8_t* encoded,
+                      size_t encoded_bytes,
+                      int16_t* decoded,
+                      int16_t* audio_type);
+
+/****************************************************************************
+ * WebRtcOpus_DecodeFec(...)
+ *
+ * This function decodes the FEC data from an Opus packet into one or more audio
+ * frames at the ACM interface's sampling rate (32 kHz).
+ *
+ * Input:
+ *      - inst               : Decoder context
+ *      - encoded            : Encoded data
+ *      - encoded_bytes      : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector (previous frame)
+ *
+ * Return value              : >0 - Samples per channel in decoded vector
+ *                              0 - No FEC data in the packet
+ *                             -1 - Error
+ */
+int WebRtcOpus_DecodeFec(OpusDecInst* inst,
+                         const uint8_t* encoded,
+                         size_t encoded_bytes,
+                         int16_t* decoded,
+                         int16_t* audio_type);
+
+/****************************************************************************
+ * WebRtcOpus_DurationEst(...)
+ *
+ * This function calculates the duration of an opus packet.
+ * Input:
+ *        - inst                 : Decoder context
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *
+ * Return value                  : The duration of the packet, in samples per
+ *                                 channel.
+ */
+int WebRtcOpus_DurationEst(OpusDecInst* inst,
+                           const uint8_t* payload,
+                           size_t payload_length_bytes);
+
+/****************************************************************************
+ * WebRtcOpus_PlcDuration(...)
+ *
+ * This function calculates the duration of a frame returned by packet loss
+ * concealment (PLC).
+ *
+ * Input:
+ *        - inst                 : Decoder context
+ *
+ * Return value                  : The duration of a frame returned by PLC, in
+ *                                 samples per channel.
+ */
+int WebRtcOpus_PlcDuration(OpusDecInst* inst);
+
+/* TODO(minyue): Check whether it is needed to add a decoder context to the
+ * arguments, like WebRtcOpus_DurationEst(...). In fact, the packet itself tells
+ * the duration. The decoder context in WebRtcOpus_DurationEst(...) is not used.
+ * So it may be advisable to remove it from WebRtcOpus_DurationEst(...). */
+
+/****************************************************************************
+ * WebRtcOpus_FecDurationEst(...)
+ *
+ * This function calculates the duration of the FEC data within an opus packet.
+ * Input:
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *        - sample_rate_hz       : Sample rate of output audio
+ *
+ * Return value                  : >0 - The duration of the FEC data in the
+ *                                 packet in samples per channel.
+ *                                  0 - No FEC data in the packet.
+ */
+int WebRtcOpus_FecDurationEst(const uint8_t* payload,
+                              size_t payload_length_bytes,
+                              int sample_rate_hz);
+
+/****************************************************************************
+ * WebRtcOpus_PacketHasFec(...)
+ *
+ * This function detects if an opus packet has FEC.
+ * Input:
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *
+ * Return value                  : 0 - the packet does NOT contain FEC.
+ *                                 1 - the packet contains FEC.
+ */
+int WebRtcOpus_PacketHasFec(const uint8_t* payload,
+                            size_t payload_length_bytes);
+
+/****************************************************************************
+ * WebRtcOpus_PacketHasVoiceActivity(...)
+ *
+ * This function returns the SILK VAD information encoded in the opus packet.
+ * For CELT-only packets that do not have VAD information, it returns -1.
+ * Input:
+ *        - payload              : Encoded data pointer
+ *        - payload_length_bytes : Bytes of encoded data
+ *
+ * Return value                  : 0 - no frame had the VAD flag set.
+ *                                 1 - at least one frame had the VAD flag set.
+ *                                -1 - VAD status could not be determined.
+ */
+int WebRtcOpus_PacketHasVoiceActivity(const uint8_t* payload,
+                                      size_t payload_length_bytes);
+
+#ifdef __cplusplus
+}  // extern "C"
+#endif
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
new file mode 100644
index 0000000000..4477e8a5f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
+
+using ::std::string;
+
+namespace webrtc {
+
+static const int kOpusBlockDurationMs = 20;
+static const int kOpusSamplingKhz = 48;
+
+class OpusSpeedTest : public AudioCodecSpeedTest {
+ protected:
+  OpusSpeedTest();
+  void SetUp() override;
+  void TearDown() override;
+  float EncodeABlock(int16_t* in_data,
+                     uint8_t* bit_stream,
+                     size_t max_bytes,
+                     size_t* encoded_bytes) override;
+  float DecodeABlock(const uint8_t* bit_stream,
+                     size_t encoded_bytes,
+                     int16_t* out_data) override;
+  WebRtcOpusEncInst* opus_encoder_;
+  WebRtcOpusDecInst* opus_decoder_;
+};
+
+OpusSpeedTest::OpusSpeedTest()
+    : AudioCodecSpeedTest(kOpusBlockDurationMs,
+                          kOpusSamplingKhz,
+                          kOpusSamplingKhz),
+      opus_encoder_(NULL),
+      opus_decoder_(NULL) {}
+
+void OpusSpeedTest::SetUp() {
+  AudioCodecSpeedTest::SetUp();
+  // If channels_ == 1, use Opus VOIP mode, otherwise, audio mode.
+  int app = channels_ == 1 ? 0 : 1;
+  /* Create encoder memory. */
+  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, app, 48000));
+  EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_, 48000));
+  /* Set bitrate. */
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_));
+}
+
+void OpusSpeedTest::TearDown() {
+  AudioCodecSpeedTest::TearDown();
+  /* Free memory. */
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+float OpusSpeedTest::EncodeABlock(int16_t* in_data,
+                                  uint8_t* bit_stream,
+                                  size_t max_bytes,
+                                  size_t* encoded_bytes) {
+  clock_t clocks = clock();
+  int value = WebRtcOpus_Encode(opus_encoder_, in_data, input_length_sample_,
+                                max_bytes, bit_stream);
+  clocks = clock() - clocks;
+  EXPECT_GT(value, 0);
+  *encoded_bytes = static_cast<size_t>(value);
+  return 1000.0 * clocks / CLOCKS_PER_SEC;
+}
+
+float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
+                                  size_t encoded_bytes,
+                                  int16_t* out_data) {
+  int value;
+  int16_t audio_type;
+  clock_t clocks = clock();
+  value = WebRtcOpus_Decode(opus_decoder_, bit_stream, encoded_bytes, out_data,
+                            &audio_type);
+  clocks = clock() - clocks;
+  EXPECT_EQ(output_length_sample_, static_cast<size_t>(value));
+  return 1000.0 * clocks / CLOCKS_PER_SEC;
+}
+
+/* Test audio length in second. */
+constexpr size_t kDurationSec = 400;
+
+#define ADD_TEST(complexity)                                           \
+  TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) {           \
+    /* Set complexity. */                                              \
+    printf("Setting complexity to %d ...\n", complexity);              \
+    EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
+    EncodeDecode(kDurationSec);                                        \
+  }
+
+ADD_TEST(10)
+ADD_TEST(9)
+ADD_TEST(8)
+ADD_TEST(7)
+ADD_TEST(6)
+ADD_TEST(5)
+ADD_TEST(4)
+ADD_TEST(3)
+ADD_TEST(2)
+ADD_TEST(1)
+ADD_TEST(0)
+
+#define ADD_BANDWIDTH_TEST(bandwidth)                                \
+  TEST_P(OpusSpeedTest, OpusSetBandwidthTest##bandwidth) {           \
+    /* Set bandwidth. */                                             \
+    printf("Setting bandwidth to %d ...\n", bandwidth);              \
+    EXPECT_EQ(0, WebRtcOpus_SetBandwidth(opus_encoder_, bandwidth)); \
+    EncodeDecode(kDurationSec);                                      \
+  }
+
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_NARROWBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_MEDIUMBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_WIDEBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_SUPERWIDEBAND)
+ADD_BANDWIDTH_TEST(OPUS_BANDWIDTH_FULLBAND)
+
+// List all test cases: (channel, bit rat, filename, extension).
+const coding_param param_set[] = {
+    std::make_tuple(1,
+                    64000,
+                    string("audio_coding/speech_mono_32_48kHz"),
+                    string("pcm"),
+                    true),
+    std::make_tuple(1,
+                    32000,
+                    string("audio_coding/speech_mono_32_48kHz"),
+                    string("pcm"),
+                    true),
+    std::make_tuple(2,
+                    64000,
+                    string("audio_coding/music_stereo_48kHz"),
+                    string("pcm"),
+                    true)};
+
+INSTANTIATE_TEST_SUITE_P(AllTest,
+                         OpusSpeedTest,
+                         ::testing::ValuesIn(param_set));
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
new file mode 100644
index 0000000000..4a9156ad58
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -0,0 +1,979 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <string>
+
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+namespace {
+// Equivalent to SDP params
+// {{"channel_mapping", "0,1,2,3"}, {"coupled_streams", "2"}}.
+constexpr unsigned char kQuadChannelMapping[] = {0, 1, 2, 3};
+constexpr int kQuadTotalStreams = 2;
+constexpr int kQuadCoupledStreams = 2;
+
+constexpr unsigned char kStereoChannelMapping[] = {0, 1};
+constexpr int kStereoTotalStreams = 1;
+constexpr int kStereoCoupledStreams = 1;
+
+constexpr unsigned char kMonoChannelMapping[] = {0};
+constexpr int kMonoTotalStreams = 1;
+constexpr int kMonoCoupledStreams = 0;
+
+void CreateSingleOrMultiStreamEncoder(WebRtcOpusEncInst** opus_encoder,
+                                      int channels,
+                                      int application,
+                                      bool use_multistream,
+                                      int encoder_sample_rate_hz) {
+  EXPECT_TRUE(channels == 1 || channels == 2 || use_multistream);
+  if (use_multistream) {
+    EXPECT_EQ(encoder_sample_rate_hz, 48000);
+    if (channels == 1) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
+                       opus_encoder, channels, application, kMonoTotalStreams,
+                       kMonoCoupledStreams, kMonoChannelMapping));
+    } else if (channels == 2) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
+                       opus_encoder, channels, application, kStereoTotalStreams,
+                       kStereoCoupledStreams, kStereoChannelMapping));
+    } else if (channels == 4) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamEncoderCreate(
+                       opus_encoder, channels, application, kQuadTotalStreams,
+                       kQuadCoupledStreams, kQuadChannelMapping));
+    } else {
+      EXPECT_TRUE(false) << channels;
+    }
+  } else {
+    EXPECT_EQ(0, WebRtcOpus_EncoderCreate(opus_encoder, channels, application,
+                                          encoder_sample_rate_hz));
+  }
+}
+
+void CreateSingleOrMultiStreamDecoder(WebRtcOpusDecInst** opus_decoder,
+                                      int channels,
+                                      bool use_multistream,
+                                      int decoder_sample_rate_hz) {
+  EXPECT_TRUE(channels == 1 || channels == 2 || use_multistream);
+  if (use_multistream) {
+    EXPECT_EQ(decoder_sample_rate_hz, 48000);
+    if (channels == 1) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
+                       opus_decoder, channels, kMonoTotalStreams,
+                       kMonoCoupledStreams, kMonoChannelMapping));
+    } else if (channels == 2) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
+                       opus_decoder, channels, kStereoTotalStreams,
+                       kStereoCoupledStreams, kStereoChannelMapping));
+    } else if (channels == 4) {
+      EXPECT_EQ(0, WebRtcOpus_MultistreamDecoderCreate(
+                       opus_decoder, channels, kQuadTotalStreams,
+                       kQuadCoupledStreams, kQuadChannelMapping));
+    } else {
+      EXPECT_TRUE(false) << channels;
+    }
+  } else {
+    EXPECT_EQ(0, WebRtcOpus_DecoderCreate(opus_decoder, channels,
+                                          decoder_sample_rate_hz));
+  }
+}
+
+int SamplesPerChannel(int sample_rate_hz, int duration_ms) {
+  const int samples_per_ms = rtc::CheckedDivExact(sample_rate_hz, 1000);
+  return samples_per_ms * duration_ms;
+}
+
+using test::AudioLoop;
+using ::testing::Combine;
+using ::testing::TestWithParam;
+using ::testing::Values;
+
+// Maximum number of bytes in output bitstream.
+const size_t kMaxBytes = 2000;
+
+class OpusTest
+    : public TestWithParam<::testing::tuple<size_t, int, bool, int, int>> {
+ protected:
+  OpusTest() = default;
+
+  void TestDtxEffect(bool dtx, int block_length_ms);
+
+  void TestCbrEffect(bool dtx, int block_length_ms);
+
+  // Prepare `speech_data_` for encoding, read from a hard-coded file.
+  // After preparation, `speech_data_.GetNextBlock()` returns a pointer to a
+  // block of `block_length_ms` milliseconds. The data is looped every
+  // `loop_length_ms` milliseconds.
+  void PrepareSpeechData(int block_length_ms, int loop_length_ms);
+
+  int EncodeDecode(WebRtcOpusEncInst* encoder,
+                   rtc::ArrayView<const int16_t> input_audio,
+                   WebRtcOpusDecInst* decoder,
+                   int16_t* output_audio,
+                   int16_t* audio_type);
+
+  void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
+                          opus_int32 expect,
+                          int32_t set);
+
+  void CheckAudioBounded(const int16_t* audio,
+                         size_t samples,
+                         size_t channels,
+                         uint16_t bound) const;
+
+  WebRtcOpusEncInst* opus_encoder_ = nullptr;
+  WebRtcOpusDecInst* opus_decoder_ = nullptr;
+  AudioLoop speech_data_;
+  uint8_t bitstream_[kMaxBytes];
+  size_t encoded_bytes_ = 0;
+  const size_t channels_{std::get<0>(GetParam())};
+  const int application_{std::get<1>(GetParam())};
+  const bool use_multistream_{std::get<2>(GetParam())};
+  const int encoder_sample_rate_hz_{std::get<3>(GetParam())};
+  const int decoder_sample_rate_hz_{std::get<4>(GetParam())};
+};
+
+}  // namespace
+
+// Singlestream: Try all combinations.
+INSTANTIATE_TEST_SUITE_P(Singlestream,
+                         OpusTest,
+                         testing::Combine(testing::Values(1, 2),
+                                          testing::Values(0, 1),
+                                          testing::Values(false),
+                                          testing::Values(16000, 48000),
+                                          testing::Values(16000, 48000)));
+
+// Multistream: Some representative cases (only 48 kHz for now).
+INSTANTIATE_TEST_SUITE_P(
+    Multistream,
+    OpusTest,
+    testing::Values(std::make_tuple(1, 0, true, 48000, 48000),
+                    std::make_tuple(2, 1, true, 48000, 48000),
+                    std::make_tuple(4, 0, true, 48000, 48000),
+                    std::make_tuple(4, 1, true, 48000, 48000)));
+
+void OpusTest::PrepareSpeechData(int block_length_ms, int loop_length_ms) {
+  std::map<int, std::string> channel_to_basename = {
+      {1, "audio_coding/testfile32kHz"},
+      {2, "audio_coding/teststereo32kHz"},
+      {4, "audio_coding/speech_4_channels_48k_one_second"}};
+  std::map<int, std::string> channel_to_suffix = {
+      {1, "pcm"}, {2, "pcm"}, {4, "wav"}};
+  const std::string file_name = webrtc::test::ResourcePath(
+      channel_to_basename[channels_], channel_to_suffix[channels_]);
+  if (loop_length_ms < block_length_ms) {
+    loop_length_ms = block_length_ms;
+  }
+  const int sample_rate_khz =
+      rtc::CheckedDivExact(encoder_sample_rate_hz_, 1000);
+  EXPECT_TRUE(speech_data_.Init(file_name,
+                                loop_length_ms * sample_rate_khz * channels_,
+                                block_length_ms * sample_rate_khz * channels_));
+}
+
+void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
+                                  opus_int32 expect,
+                                  int32_t set) {
+  opus_int32 bandwidth;
+  EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, set));
+  EXPECT_EQ(0, WebRtcOpus_GetMaxPlaybackRate(opus_encoder_, &bandwidth));
+  EXPECT_EQ(expect, bandwidth);
+}
+
+void OpusTest::CheckAudioBounded(const int16_t* audio,
+                                 size_t samples,
+                                 size_t channels,
+                                 uint16_t bound) const {
+  for (size_t i = 0; i < samples; ++i) {
+    for (size_t c = 0; c < channels; ++c) {
+      ASSERT_GE(audio[i * channels + c], -bound);
+      ASSERT_LE(audio[i * channels + c], bound);
+    }
+  }
+}
+
+int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
+                           rtc::ArrayView<const int16_t> input_audio,
+                           WebRtcOpusDecInst* decoder,
+                           int16_t* output_audio,
+                           int16_t* audio_type) {
+  const int input_samples_per_channel =
+      rtc::CheckedDivExact(input_audio.size(), channels_);
+  int encoded_bytes_int =
+      WebRtcOpus_Encode(encoder, input_audio.data(), input_samples_per_channel,
+                        kMaxBytes, bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
+  encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
+  if (encoded_bytes_ != 0) {
+    int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_);
+    int act_len = WebRtcOpus_Decode(decoder, bitstream_, encoded_bytes_,
+                                    output_audio, audio_type);
+    EXPECT_EQ(est_len, act_len);
+    return act_len;
+  } else {
+    int total_dtx_len = 0;
+    const int output_samples_per_channel = input_samples_per_channel *
+                                           decoder_sample_rate_hz_ /
+                                           encoder_sample_rate_hz_;
+    while (total_dtx_len < output_samples_per_channel) {
+      int est_len = WebRtcOpus_DurationEst(decoder, NULL, 0);
+      int act_len = WebRtcOpus_Decode(decoder, NULL, 0,
+                                      &output_audio[total_dtx_len * channels_],
+                                      audio_type);
+      EXPECT_EQ(est_len, act_len);
+      total_dtx_len += act_len;
+    }
+    return total_dtx_len;
+  }
+}
+
+// Test if encoder/decoder can enter DTX mode properly and do not enter DTX when
+// they should not. This test is signal dependent.
+void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
+  PrepareSpeechData(block_length_ms, 2000);
+  const size_t input_samples =
+      rtc::CheckedDivExact(encoder_sample_rate_hz_, 1000) * block_length_ms;
+  const size_t output_samples =
+      rtc::CheckedDivExact(decoder_sample_rate_hz_, 1000) * block_length_ms;
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Set input audio as silence.
+  std::vector<int16_t> silence(input_samples * channels_, 0);
+
+  // Setting DTX.
+  EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_)
+                   : WebRtcOpus_DisableDtx(opus_encoder_));
+
+  int16_t audio_type;
+  int16_t* output_data_decode = new int16_t[output_samples * channels_];
+
+  for (int i = 0; i < 100; ++i) {
+    EXPECT_EQ(output_samples,
+              static_cast<size_t>(EncodeDecode(
+                  opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+                  output_data_decode, &audio_type)));
+    // If not DTX, it should never enter DTX mode. If DTX, we do not care since
+    // whether it enters DTX depends on the signal type.
+    if (!dtx) {
+      EXPECT_GT(encoded_bytes_, 1U);
+      EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(0, audio_type);  // Speech.
+    }
+  }
+
+  // We input some silent segments. In DTX mode, the encoder will stop sending.
+  // However, DTX may happen after a while.
+  for (int i = 0; i < 30; ++i) {
+    EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                  opus_encoder_, silence, opus_decoder_,
+                                  output_data_decode, &audio_type)));
+    if (!dtx) {
+      EXPECT_GT(encoded_bytes_, 1U);
+      EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(0, audio_type);  // Speech.
+    } else if (encoded_bytes_ == 1) {
+      EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(2, audio_type);  // Comfort noise.
+      break;
+    }
+  }
+
+  // When Opus is in DTX, it wakes up in a regular basis. It sends two packets,
+  // one with an arbitrary size and the other of 1-byte, then stops sending for
+  // a certain number of frames.
+
+  // `max_dtx_frames` is the maximum number of frames Opus can stay in DTX.
+  // TODO(kwiberg): Why does this number depend on the encoding sample rate?
+  const int max_dtx_frames =
+      (encoder_sample_rate_hz_ == 16000 ? 800 : 400) / block_length_ms + 1;
+
+  // We run `kRunTimeMs` milliseconds of pure silence.
+  const int kRunTimeMs = 4500;
+
+  // We check that, after a `kCheckTimeMs` milliseconds (given that the CNG in
+  // Opus needs time to adapt), the absolute values of DTX decoded signal are
+  // bounded by `kOutputValueBound`.
+  const int kCheckTimeMs = 4000;
+
+#if defined(OPUS_FIXED_POINT)
+  // Fixed-point Opus generates a random (comfort) noise, which has a less
+  // predictable value bound than its floating-point Opus. This value depends on
+  // input signal, and the time window for checking the output values (between
+  // `kCheckTimeMs` and `kRunTimeMs`).
+  const uint16_t kOutputValueBound = 30;
+
+#else
+  const uint16_t kOutputValueBound = 2;
+#endif
+
+  int time = 0;
+  while (time < kRunTimeMs) {
+    // DTX mode is maintained for maximum `max_dtx_frames` frames.
+    int i = 0;
+    for (; i < max_dtx_frames; ++i) {
+      time += block_length_ms;
+      EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                    opus_encoder_, silence, opus_decoder_,
+                                    output_data_decode, &audio_type)));
+      if (dtx) {
+        if (encoded_bytes_ > 1)
+          break;
+        EXPECT_EQ(0U, encoded_bytes_)  // Send 0 byte.
+            << "Opus should have entered DTX mode.";
+        EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
+        EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
+        EXPECT_EQ(2, audio_type);  // Comfort noise.
+        if (time >= kCheckTimeMs) {
+          CheckAudioBounded(output_data_decode, output_samples, channels_,
+                            kOutputValueBound);
+        }
+      } else {
+        EXPECT_GT(encoded_bytes_, 1U);
+        EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+        EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+        EXPECT_EQ(0, audio_type);  // Speech.
+      }
+    }
+
+    if (dtx) {
+      // With DTX, Opus must stop transmission for some time.
+      EXPECT_GT(i, 1);
+    }
+
+    // We expect a normal payload.
+    EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+    EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+    EXPECT_EQ(0, audio_type);  // Speech.
+
+    // Enters DTX again immediately.
+    time += block_length_ms;
+    EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                  opus_encoder_, silence, opus_decoder_,
+                                  output_data_decode, &audio_type)));
+    if (dtx) {
+      EXPECT_EQ(1U, encoded_bytes_);  // Send 1 byte.
+      EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(1, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(2, audio_type);  // Comfort noise.
+      if (time >= kCheckTimeMs) {
+        CheckAudioBounded(output_data_decode, output_samples, channels_,
+                          kOutputValueBound);
+      }
+    } else {
+      EXPECT_GT(encoded_bytes_, 1U);
+      EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+      EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+      EXPECT_EQ(0, audio_type);  // Speech.
+    }
+  }
+
+  silence[0] = 10000;
+  if (dtx) {
+    // Verify that encoder/decoder can jump out from DTX mode.
+    EXPECT_EQ(output_samples, static_cast<size_t>(EncodeDecode(
+                                  opus_encoder_, silence, opus_decoder_,
+                                  output_data_decode, &audio_type)));
+    EXPECT_GT(encoded_bytes_, 1U);
+    EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
+    EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
+    EXPECT_EQ(0, audio_type);  // Speech.
+  }
+
+  // Free memory.
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+// Test if CBR does what we expect.
+void OpusTest::TestCbrEffect(bool cbr, int block_length_ms) {
+  PrepareSpeechData(block_length_ms, 2000);
+  const size_t output_samples =
+      rtc::CheckedDivExact(decoder_sample_rate_hz_, 1000) * block_length_ms;
+
+  int32_t max_pkt_size_diff = 0;
+  int32_t prev_pkt_size = 0;
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Setting CBR.
+  EXPECT_EQ(0, cbr ? WebRtcOpus_EnableCbr(opus_encoder_)
+                   : WebRtcOpus_DisableCbr(opus_encoder_));
+
+  int16_t audio_type;
+  std::vector<int16_t> audio_out(output_samples * channels_);
+  for (int i = 0; i < 100; ++i) {
+    EXPECT_EQ(output_samples,
+              static_cast<size_t>(
+                  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                               opus_decoder_, audio_out.data(), &audio_type)));
+
+    if (prev_pkt_size > 0) {
+      int32_t diff = std::abs((int32_t)encoded_bytes_ - prev_pkt_size);
+      max_pkt_size_diff = std::max(max_pkt_size_diff, diff);
+    }
+    prev_pkt_size = rtc::checked_cast<int32_t>(encoded_bytes_);
+  }
+
+  if (cbr) {
+    EXPECT_EQ(max_pkt_size_diff, 0);
+  } else {
+    EXPECT_GT(max_pkt_size_diff, 0);
+  }
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+// Test failing Create.
+TEST(OpusTest, OpusCreateFail) {
+  WebRtcOpusEncInst* opus_encoder;
+  WebRtcOpusDecInst* opus_decoder;
+
+  // Test to see that an invalid pointer is caught.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(NULL, 1, 0, 48000));
+  // Invalid channel number.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 257, 0, 48000));
+  // Invalid applciation mode.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 2, 48000));
+  // Invalid sample rate.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderCreate(&opus_encoder, 1, 0, 12345));
+
+  EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(NULL, 1, 48000));
+  // Invalid channel number.
+  EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 257, 48000));
+  // Invalid sample rate.
+  EXPECT_EQ(-1, WebRtcOpus_DecoderCreate(&opus_decoder, 1, 12345));
+}
+
+// Test failing Free.
+TEST(OpusTest, OpusFreeFail) {
+  // Test to see that an invalid pointer is caught.
+  EXPECT_EQ(-1, WebRtcOpus_EncoderFree(NULL));
+  EXPECT_EQ(-1, WebRtcOpus_DecoderFree(NULL));
+}
+
+// Test normal Create and Free.
+TEST_P(OpusTest, OpusCreateFree) {
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+  EXPECT_TRUE(opus_encoder_ != NULL);
+  EXPECT_TRUE(opus_decoder_ != NULL);
+  // Free encoder and decoder memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+#define ENCODER_CTL(inst, vargs)               \
+  inst->encoder                                \
+      ? opus_encoder_ctl(inst->encoder, vargs) \
+      : opus_multistream_encoder_ctl(inst->multistream_encoder, vargs)
+
+TEST_P(OpusTest, OpusEncodeDecode) {
+  PrepareSpeechData(20, 20);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Check number of channels for decoder.
+  EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
+
+  // Check application mode.
+  opus_int32 app;
+  ENCODER_CTL(opus_encoder_, OPUS_GET_APPLICATION(&app));
+  EXPECT_EQ(application_ == 0 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO,
+            app);
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  int16_t* output_data_decode =
+      new int16_t[decode_samples_per_channel * channels_];
+  EXPECT_EQ(decode_samples_per_channel,
+            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                         opus_decoder_, output_data_decode, &audio_type));
+
+  // Free memory.
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusSetBitRate) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
+
+  // Create encoder memory, try with different bitrates.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 30000));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 300000));
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 600000));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusSetComplexity) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 9));
+
+  // Create encoder memory, try with different complexities.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 0));
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 10));
+  EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 11));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusSetBandwidth) {
+  if (channels_ > 2) {
+    // TODO(webrtc:10217): investigate why multi-stream Opus reports
+    // narrowband when it's configured with FULLBAND.
+    return;
+  }
+  PrepareSpeechData(20, 20);
+
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  std::unique_ptr<int16_t[]> output_data_decode(
+      new int16_t[decode_samples_per_channel * channels_]());
+
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1,
+            WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND));
+  EXPECT_EQ(-1, WebRtcOpus_GetBandwidth(opus_encoder_));
+
+  // Create encoder memory, try with different bandwidths.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  EXPECT_EQ(-1, WebRtcOpus_SetBandwidth(opus_encoder_,
+                                        OPUS_BANDWIDTH_NARROWBAND - 1));
+  EXPECT_EQ(0,
+            WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND));
+  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+               output_data_decode.get(), &audio_type);
+  EXPECT_EQ(OPUS_BANDWIDTH_NARROWBAND, WebRtcOpus_GetBandwidth(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_FULLBAND));
+  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+               output_data_decode.get(), &audio_type);
+  EXPECT_EQ(encoder_sample_rate_hz_ == 16000 ? OPUS_BANDWIDTH_WIDEBAND
+                                             : OPUS_BANDWIDTH_FULLBAND,
+            WebRtcOpus_GetBandwidth(opus_encoder_));
+  EXPECT_EQ(
+      -1, WebRtcOpus_SetBandwidth(opus_encoder_, OPUS_BANDWIDTH_FULLBAND + 1));
+  EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
+               output_data_decode.get(), &audio_type);
+  EXPECT_EQ(encoder_sample_rate_hz_ == 16000 ? OPUS_BANDWIDTH_WIDEBAND
+                                             : OPUS_BANDWIDTH_FULLBAND,
+            WebRtcOpus_GetBandwidth(opus_encoder_));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusForceChannels) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
+
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  ASSERT_NE(nullptr, opus_encoder_);
+
+  if (channels_ >= 2) {
+    EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 3));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 2));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 0));
+  } else {
+    EXPECT_EQ(-1, WebRtcOpus_SetForceChannels(opus_encoder_, 2));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 1));
+    EXPECT_EQ(0, WebRtcOpus_SetForceChannels(opus_encoder_, 0));
+  }
+
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+// Encode and decode one frame, initialize the decoder and
+// decode once more.
+TEST_P(OpusTest, OpusDecodeInit) {
+  PrepareSpeechData(20, 20);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  int16_t* output_data_decode =
+      new int16_t[decode_samples_per_channel * channels_];
+  EXPECT_EQ(decode_samples_per_channel,
+            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                         opus_decoder_, output_data_decode, &audio_type));
+
+  WebRtcOpus_DecoderInit(opus_decoder_);
+
+  EXPECT_EQ(decode_samples_per_channel,
+            WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
+                              output_data_decode, &audio_type));
+
+  // Free memory.
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusEnableDisableFec) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_EnableFec(opus_encoder_));
+  EXPECT_EQ(-1, WebRtcOpus_DisableFec(opus_encoder_));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusEnableDisableDtx) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_EnableDtx(opus_encoder_));
+  EXPECT_EQ(-1, WebRtcOpus_DisableDtx(opus_encoder_));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  opus_int32 dtx;
+
+  // DTX is off by default.
+  ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
+  EXPECT_EQ(0, dtx);
+
+  // Test to enable DTX.
+  EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
+  ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
+  EXPECT_EQ(1, dtx);
+
+  // Test to disable DTX.
+  EXPECT_EQ(0, WebRtcOpus_DisableDtx(opus_encoder_));
+  ENCODER_CTL(opus_encoder_, OPUS_GET_DTX(&dtx));
+  EXPECT_EQ(0, dtx);
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusDtxOff) {
+  TestDtxEffect(false, 10);
+  TestDtxEffect(false, 20);
+  TestDtxEffect(false, 40);
+}
+
+TEST_P(OpusTest, OpusDtxOn) {
+  if (channels_ > 2 || application_ != 0) {
+    // DTX does not work with OPUS_APPLICATION_AUDIO at low complexity settings.
+    // TODO(webrtc:10218): adapt the test to the sizes and order of multi-stream
+    // DTX packets.
+    return;
+  }
+  TestDtxEffect(true, 10);
+  TestDtxEffect(true, 20);
+  TestDtxEffect(true, 40);
+}
+
+TEST_P(OpusTest, OpusCbrOff) {
+  TestCbrEffect(false, 10);
+  TestCbrEffect(false, 20);
+  TestCbrEffect(false, 40);
+}
+
+TEST_P(OpusTest, OpusCbrOn) {
+  TestCbrEffect(true, 10);
+  TestCbrEffect(true, 20);
+  TestCbrEffect(true, 40);
+}
+
+TEST_P(OpusTest, OpusSetPacketLossRate) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
+  EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, -1));
+  EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 101));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+TEST_P(OpusTest, OpusSetMaxPlaybackRate) {
+  // Test without creating encoder memory.
+  EXPECT_EQ(-1, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, 20000));
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 48000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 24001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_SUPERWIDEBAND, 24000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_SUPERWIDEBAND, 16001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_WIDEBAND, 16000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_WIDEBAND, 12001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_MEDIUMBAND, 12000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_MEDIUMBAND, 8001);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND, 8000);
+  SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_NARROWBAND, 4000);
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+}
+
+// Test PLC.
+TEST_P(OpusTest, OpusDecodePlc) {
+  PrepareSpeechData(20, 20);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Check number of channels for decoder.
+  EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  int16_t* output_data_decode =
+      new int16_t[decode_samples_per_channel * channels_];
+  EXPECT_EQ(decode_samples_per_channel,
+            EncodeDecode(opus_encoder_, speech_data_.GetNextBlock(),
+                         opus_decoder_, output_data_decode, &audio_type));
+
+  // Call decoder PLC.
+  constexpr int kPlcDurationMs = 10;
+  const int plc_samples = decoder_sample_rate_hz_ * kPlcDurationMs / 1000;
+  int16_t* plc_buffer = new int16_t[plc_samples * channels_];
+  EXPECT_EQ(plc_samples,
+            WebRtcOpus_Decode(opus_decoder_, NULL, 0, plc_buffer, &audio_type));
+
+  // Free memory.
+  delete[] plc_buffer;
+  delete[] output_data_decode;
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+// Duration estimation.
+TEST_P(OpusTest, OpusDurationEstimation) {
+  PrepareSpeechData(20, 20);
+
+  // Create.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+
+  // 10 ms. We use only first 10 ms of a 20 ms block.
+  auto speech_block = speech_data_.GetNextBlock();
+  int encoded_bytes_int = WebRtcOpus_Encode(
+      opus_encoder_, speech_block.data(),
+      rtc::CheckedDivExact(speech_block.size(), 2 * channels_), kMaxBytes,
+      bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
+  EXPECT_EQ(SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/10),
+            WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
+                                   static_cast<size_t>(encoded_bytes_int)));
+
+  // 20 ms
+  speech_block = speech_data_.GetNextBlock();
+  encoded_bytes_int =
+      WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+                        rtc::CheckedDivExact(speech_block.size(), channels_),
+                        kMaxBytes, bitstream_);
+  EXPECT_GE(encoded_bytes_int, 0);
+  EXPECT_EQ(SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20),
+            WebRtcOpus_DurationEst(opus_decoder_, bitstream_,
+                                   static_cast<size_t>(encoded_bytes_int)));
+
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST_P(OpusTest, OpusDecodeRepacketized) {
+  if (channels_ > 2) {
+    // As per the Opus documentation
+    // https://mf4.xiph.org/jenkins/view/opus/job/opus/ws/doc/html/group__opus__repacketizer.html#details,
+    // multiple streams are not supported.
+    return;
+  }
+  constexpr size_t kPackets = 6;
+
+  PrepareSpeechData(20, 20 * kPackets);
+
+  // Create encoder memory.
+  CreateSingleOrMultiStreamEncoder(&opus_encoder_, channels_, application_,
+                                   use_multistream_, encoder_sample_rate_hz_);
+  ASSERT_NE(nullptr, opus_encoder_);
+  CreateSingleOrMultiStreamDecoder(&opus_decoder_, channels_, use_multistream_,
+                                   decoder_sample_rate_hz_);
+  ASSERT_NE(nullptr, opus_decoder_);
+
+  // Set bitrate.
+  EXPECT_EQ(
+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
+
+  // Check number of channels for decoder.
+  EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
+
+  // Encode & decode.
+  int16_t audio_type;
+  const int decode_samples_per_channel =
+      SamplesPerChannel(decoder_sample_rate_hz_, /*ms=*/20);
+  std::unique_ptr<int16_t[]> output_data_decode(
+      new int16_t[kPackets * decode_samples_per_channel * channels_]);
+  OpusRepacketizer* rp = opus_repacketizer_create();
+
+  size_t num_packets = 0;
+  constexpr size_t kMaxCycles = 100;
+  for (size_t idx = 0; idx < kMaxCycles; ++idx) {
+    auto speech_block = speech_data_.GetNextBlock();
+    encoded_bytes_ =
+        WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
+                          rtc::CheckedDivExact(speech_block.size(), channels_),
+                          kMaxBytes, bitstream_);
+    if (opus_repacketizer_cat(rp, bitstream_,
+                              rtc::checked_cast<opus_int32>(encoded_bytes_)) ==
+        OPUS_OK) {
+      ++num_packets;
+      if (num_packets == kPackets) {
+        break;
+      }
+    } else {
+      // Opus repacketizer cannot guarantee a success. We try again if it fails.
+      opus_repacketizer_init(rp);
+      num_packets = 0;
+    }
+  }
+  EXPECT_EQ(kPackets, num_packets);
+
+  encoded_bytes_ = opus_repacketizer_out(rp, bitstream_, kMaxBytes);
+
+  EXPECT_EQ(decode_samples_per_channel * kPackets,
+            static_cast<size_t>(WebRtcOpus_DurationEst(
+                opus_decoder_, bitstream_, encoded_bytes_)));
+
+  EXPECT_EQ(decode_samples_per_channel * kPackets,
+            static_cast<size_t>(
+                WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
+                                  output_data_decode.get(), &audio_type)));
+
+  // Free memory.
+  opus_repacketizer_destroy(rp);
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
+}
+
+TEST(OpusVadTest, CeltUnknownStatus) {
+  const uint8_t celt[] = {0x80};
+  EXPECT_EQ(WebRtcOpus_PacketHasVoiceActivity(celt, 1), -1);
+}
+
+TEST(OpusVadTest, Mono20msVadSet) {
+  uint8_t silk20msMonoVad[] = {0x78, 0x80};
+  EXPECT_TRUE(WebRtcOpus_PacketHasVoiceActivity(silk20msMonoVad, 2));
+}
+
+TEST(OpusVadTest, Mono20MsVadUnset) {
+  uint8_t silk20msMonoSilence[] = {0x78, 0x00};
+  EXPECT_FALSE(WebRtcOpus_PacketHasVoiceActivity(silk20msMonoSilence, 2));
+}
+
+TEST(OpusVadTest, Stereo20MsVadOnSideChannel) {
+  uint8_t silk20msStereoVadSideChannel[] = {0x78 | 0x04, 0x20};
+  EXPECT_TRUE(
+      WebRtcOpus_PacketHasVoiceActivity(silk20msStereoVadSideChannel, 2));
+}
+
+TEST(OpusVadTest, TwoOpusMonoFramesVadOnSecond) {
+  uint8_t twoMonoFrames[] = {0x78 | 0x1, 0x00, 0x80};
+  EXPECT_TRUE(WebRtcOpus_PacketHasVoiceActivity(twoMonoFrames, 3));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn
new file mode 100644
index 0000000000..8bc0bf5e0e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/BUILD.gn
@@ -0,0 +1,55 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../../../webrtc.gni")
+
+visibility = [
+  ":*",
+  "../../../:*",
+]
+
+if (rtc_include_tests) {
+  rtc_library("test") {
+    testonly = true
+
+    sources = [
+      "audio_ring_buffer.cc",
+      "audio_ring_buffer.h",
+      "blocker.cc",
+      "blocker.h",
+      "lapped_transform.cc",
+      "lapped_transform.h",
+    ]
+
+    deps = [
+      "../../../../../common_audio",
+      "../../../../../common_audio:common_audio_c",
+      "../../../../../rtc_base:checks",
+      "../../../../../rtc_base/memory:aligned_malloc",
+    ]
+  }
+
+  rtc_library("test_unittest") {
+    testonly = true
+
+    sources = [
+      "audio_ring_buffer_unittest.cc",
+      "blocker_unittest.cc",
+      "lapped_transform_unittest.cc",
+    ]
+
+    deps = [
+      ":test",
+      "../../../../../common_audio",
+      "../../../../../common_audio:common_audio_c",
+      "../../../../../rtc_base:macromagic",
+      "../../../../../test:test_support",
+      "//testing/gtest",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc
new file mode 100644
index 0000000000..2a71b43d2c
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.cc
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+
+#include "common_audio/ring_buffer.h"
+#include "rtc_base/checks.h"
+
+// This is a simple multi-channel wrapper over the ring_buffer.h C interface.
+
+namespace webrtc {
+
+AudioRingBuffer::AudioRingBuffer(size_t channels, size_t max_frames) {
+  buffers_.reserve(channels);
+  for (size_t i = 0; i < channels; ++i)
+    buffers_.push_back(WebRtc_CreateBuffer(max_frames, sizeof(float)));
+}
+
+AudioRingBuffer::~AudioRingBuffer() {
+  for (auto* buf : buffers_)
+    WebRtc_FreeBuffer(buf);
+}
+
+void AudioRingBuffer::Write(const float* const* data,
+                            size_t channels,
+                            size_t frames) {
+  RTC_DCHECK_EQ(buffers_.size(), channels);
+  for (size_t i = 0; i < channels; ++i) {
+    const size_t written = WebRtc_WriteBuffer(buffers_[i], data[i], frames);
+    RTC_CHECK_EQ(written, frames);
+  }
+}
+
+void AudioRingBuffer::Read(float* const* data, size_t channels, size_t frames) {
+  RTC_DCHECK_EQ(buffers_.size(), channels);
+  for (size_t i = 0; i < channels; ++i) {
+    const size_t read =
+        WebRtc_ReadBuffer(buffers_[i], nullptr, data[i], frames);
+    RTC_CHECK_EQ(read, frames);
+  }
+}
+
+size_t AudioRingBuffer::ReadFramesAvailable() const {
+  // All buffers have the same amount available.
+  return WebRtc_available_read(buffers_[0]);
+}
+
+size_t AudioRingBuffer::WriteFramesAvailable() const {
+  // All buffers have the same amount available.
+  return WebRtc_available_write(buffers_[0]);
+}
+
+void AudioRingBuffer::MoveReadPositionForward(size_t frames) {
+  for (auto* buf : buffers_) {
+    const size_t moved =
+        static_cast<size_t>(WebRtc_MoveReadPtr(buf, static_cast<int>(frames)));
+    RTC_CHECK_EQ(moved, frames);
+  }
+}
+
+void AudioRingBuffer::MoveReadPositionBackward(size_t frames) {
+  for (auto* buf : buffers_) {
+    const size_t moved = static_cast<size_t>(
+        -WebRtc_MoveReadPtr(buf, -static_cast<int>(frames)));
+    RTC_CHECK_EQ(moved, frames);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h
new file mode 100644
index 0000000000..a280ca2410
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+struct RingBuffer;
+
+namespace webrtc {
+
+// A ring buffer tailored for float deinterleaved audio. Any operation that
+// cannot be performed as requested will cause a crash (e.g. insufficient data
+// in the buffer to fulfill a read request.)
+class AudioRingBuffer final {
+ public:
+  // Specify the number of channels and maximum number of frames the buffer will
+  // contain.
+  AudioRingBuffer(size_t channels, size_t max_frames);
+  ~AudioRingBuffer();
+
+  // Copies `data` to the buffer and advances the write pointer. `channels` must
+  // be the same as at creation time.
+  void Write(const float* const* data, size_t channels, size_t frames);
+
+  // Copies from the buffer to `data` and advances the read pointer. `channels`
+  // must be the same as at creation time.
+  void Read(float* const* data, size_t channels, size_t frames);
+
+  size_t ReadFramesAvailable() const;
+  size_t WriteFramesAvailable() const;
+
+  // Moves the read position. The forward version advances the read pointer
+  // towards the write pointer and the backward verison withdraws the read
+  // pointer away from the write pointer (i.e. flushing and stuffing the buffer
+  // respectively.)
+  void MoveReadPositionForward(size_t frames);
+  void MoveReadPositionBackward(size_t frames);
+
+ private:
+  // TODO(kwiberg): Use std::vector<std::unique_ptr<RingBuffer>> instead.
+  std::vector<RingBuffer*> buffers_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_AUDIO_RING_BUFFER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
new file mode 100644
index 0000000000..6dbc8ee9fe
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/audio_ring_buffer_unittest.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+
+#include <memory>
+
+#include "common_audio/channel_buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class AudioRingBufferTest
+    : public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > {
+};
+
+void ReadAndWriteTest(const ChannelBuffer<float>& input,
+                      size_t num_write_chunk_frames,
+                      size_t num_read_chunk_frames,
+                      size_t buffer_frames,
+                      ChannelBuffer<float>* output) {
+  const size_t num_channels = input.num_channels();
+  const size_t total_frames = input.num_frames();
+  AudioRingBuffer buf(num_channels, buffer_frames);
+  std::unique_ptr<float*[]> slice(new float*[num_channels]);
+
+  size_t input_pos = 0;
+  size_t output_pos = 0;
+  while (input_pos + buf.WriteFramesAvailable() < total_frames) {
+    // Write until the buffer is as full as possible.
+    while (buf.WriteFramesAvailable() >= num_write_chunk_frames) {
+      buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+                num_write_chunk_frames);
+      input_pos += num_write_chunk_frames;
+    }
+    // Read until the buffer is as empty as possible.
+    while (buf.ReadFramesAvailable() >= num_read_chunk_frames) {
+      EXPECT_LT(output_pos, total_frames);
+      buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+               num_read_chunk_frames);
+      output_pos += num_read_chunk_frames;
+    }
+  }
+
+  // Write and read the last bit.
+  if (input_pos < total_frames) {
+    buf.Write(input.Slice(slice.get(), input_pos), num_channels,
+              total_frames - input_pos);
+  }
+  if (buf.ReadFramesAvailable()) {
+    buf.Read(output->Slice(slice.get(), output_pos), num_channels,
+             buf.ReadFramesAvailable());
+  }
+  EXPECT_EQ(0u, buf.ReadFramesAvailable());
+}
+
+TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
+  const size_t kFrames = 5000;
+  const size_t num_channels = ::testing::get<3>(GetParam());
+
+  // Initialize the input data to an increasing sequence.
+  ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
+  for (size_t i = 0; i < num_channels; ++i)
+    for (size_t j = 0; j < kFrames; ++j)
+      input.channels()[i][j] = (i + 1) * (j + 1);
+
+  ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels));
+  ReadAndWriteTest(input, ::testing::get<0>(GetParam()),
+                   ::testing::get<1>(GetParam()), ::testing::get<2>(GetParam()),
+                   &output);
+
+  // Verify the read data matches the input.
+  for (size_t i = 0; i < num_channels; ++i)
+    for (size_t j = 0; j < kFrames; ++j)
+      EXPECT_EQ(input.channels()[i][j], output.channels()[i][j]);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioRingBufferTest,
+    AudioRingBufferTest,
+    ::testing::Combine(::testing::Values(10, 20, 42),  // num_write_chunk_frames
+                       ::testing::Values(1, 10, 17),   // num_read_chunk_frames
+                       ::testing::Values(100, 256),    // buffer_frames
+                       ::testing::Values(1, 4)));      // num_channels
+
+TEST_F(AudioRingBufferTest, MoveReadPosition) {
+  const size_t kNumChannels = 1;
+  const float kInputArray[] = {1, 2, 3, 4};
+  const size_t kNumFrames = sizeof(kInputArray) / sizeof(*kInputArray);
+  ChannelBuffer<float> input(kNumFrames, kNumChannels);
+  input.SetDataForTesting(kInputArray, kNumFrames);
+  AudioRingBuffer buf(kNumChannels, kNumFrames);
+  buf.Write(input.channels(), kNumChannels, kNumFrames);
+
+  buf.MoveReadPositionForward(3);
+  ChannelBuffer<float> output(1, kNumChannels);
+  buf.Read(output.channels(), kNumChannels, 1);
+  EXPECT_EQ(4, output.channels()[0][0]);
+  buf.MoveReadPositionBackward(3);
+  buf.Read(output.channels(), kNumChannels, 1);
+  EXPECT_EQ(2, output.channels()[0][0]);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc
new file mode 100644
index 0000000000..33406cead9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.cc
@@ -0,0 +1,215 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/blocker.h"
+
+#include <string.h>
+
+#include "rtc_base/checks.h"
+
+namespace {
+
+// Adds `a` and `b` frame by frame into `result` (basically matrix addition).
+void AddFrames(const float* const* a,
+               size_t a_start_index,
+               const float* const* b,
+               int b_start_index,
+               size_t num_frames,
+               size_t num_channels,
+               float* const* result,
+               size_t result_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    for (size_t j = 0; j < num_frames; ++j) {
+      result[i][j + result_start_index] =
+          a[i][j + a_start_index] + b[i][j + b_start_index];
+    }
+  }
+}
+
+// Copies `src` into `dst` channel by channel.
+void CopyFrames(const float* const* src,
+                size_t src_start_index,
+                size_t num_frames,
+                size_t num_channels,
+                float* const* dst,
+                size_t dst_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    memcpy(&dst[i][dst_start_index], &src[i][src_start_index],
+           num_frames * sizeof(dst[i][dst_start_index]));
+  }
+}
+
+// Moves `src` into `dst` channel by channel.
+void MoveFrames(const float* const* src,
+                size_t src_start_index,
+                size_t num_frames,
+                size_t num_channels,
+                float* const* dst,
+                size_t dst_start_index) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    memmove(&dst[i][dst_start_index], &src[i][src_start_index],
+            num_frames * sizeof(dst[i][dst_start_index]));
+  }
+}
+
+void ZeroOut(float* const* buffer,
+             size_t starting_idx,
+             size_t num_frames,
+             size_t num_channels) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    memset(&buffer[i][starting_idx], 0,
+           num_frames * sizeof(buffer[i][starting_idx]));
+  }
+}
+
+// Pointwise multiplies each channel of `frames` with `window`. Results are
+// stored in `frames`.
+void ApplyWindow(const float* window,
+                 size_t num_frames,
+                 size_t num_channels,
+                 float* const* frames) {
+  for (size_t i = 0; i < num_channels; ++i) {
+    for (size_t j = 0; j < num_frames; ++j) {
+      frames[i][j] = frames[i][j] * window[j];
+    }
+  }
+}
+
+size_t gcd(size_t a, size_t b) {
+  size_t tmp;
+  while (b) {
+    tmp = a;
+    a = b;
+    b = tmp % b;
+  }
+  return a;
+}
+
+}  // namespace
+
+namespace webrtc {
+
+Blocker::Blocker(size_t chunk_size,
+                 size_t block_size,
+                 size_t num_input_channels,
+                 size_t num_output_channels,
+                 const float* window,
+                 size_t shift_amount,
+                 BlockerCallback* callback)
+    : chunk_size_(chunk_size),
+      block_size_(block_size),
+      num_input_channels_(num_input_channels),
+      num_output_channels_(num_output_channels),
+      initial_delay_(block_size_ - gcd(chunk_size, shift_amount)),
+      frame_offset_(0),
+      input_buffer_(num_input_channels_, chunk_size_ + initial_delay_),
+      output_buffer_(chunk_size_ + initial_delay_, num_output_channels_),
+      input_block_(block_size_, num_input_channels_),
+      output_block_(block_size_, num_output_channels_),
+      window_(new float[block_size_]),
+      shift_amount_(shift_amount),
+      callback_(callback) {
+  RTC_CHECK_LE(num_output_channels_, num_input_channels_);
+  RTC_CHECK_LE(shift_amount_, block_size_);
+
+  memcpy(window_.get(), window, block_size_ * sizeof(*window_.get()));
+  input_buffer_.MoveReadPositionBackward(initial_delay_);
+}
+
+Blocker::~Blocker() = default;
+
+// When block_size < chunk_size the input and output buffers look like this:
+//
+//                      delay*             chunk_size    chunk_size + delay*
+//  buffer: <-------------|---------------------|---------------|>
+//                _a_              _b_                 _c_
+//
+// On each call to ProcessChunk():
+// 1. New input gets read into sections _b_ and _c_ of the input buffer.
+// 2. We block starting from frame_offset.
+// 3. We block until we reach a block `bl` that doesn't contain any frames
+//    from sections _a_ or _b_ of the input buffer.
+// 4. We window the current block, fire the callback for processing, window
+//    again, and overlap/add to the output buffer.
+// 5. We copy sections _a_ and _b_ of the output buffer into output.
+// 6. For both the input and the output buffers, we copy section _c_ into
+//    section _a_.
+// 7. We set the new frame_offset to be the difference between the first frame
+//    of `bl` and the border between sections _b_ and _c_.
+//
+// When block_size > chunk_size the input and output buffers look like this:
+//
+//                   chunk_size               delay*       chunk_size + delay*
+//  buffer: <-------------|---------------------|---------------|>
+//                _a_              _b_                 _c_
+//
+// On each call to ProcessChunk():
+// The procedure is the same as above, except for:
+// 1. New input gets read into section _c_ of the input buffer.
+// 3. We block until we reach a block `bl` that doesn't contain any frames
+//    from section _a_ of the input buffer.
+// 5. We copy section _a_ of the output buffer into output.
+// 6. For both the input and the output buffers, we copy sections _b_ and _c_
+//    into section _a_ and _b_.
+// 7. We set the new frame_offset to be the difference between the first frame
+//    of `bl` and the border between sections _a_ and _b_.
+//
+// * delay here refers to inintial_delay_
+//
+// TODO(claguna): Look at using ring buffers to eliminate some copies.
+void Blocker::ProcessChunk(const float* const* input,
+                           size_t chunk_size,
+                           size_t num_input_channels,
+                           size_t num_output_channels,
+                           float* const* output) {
+  RTC_CHECK_EQ(chunk_size, chunk_size_);
+  RTC_CHECK_EQ(num_input_channels, num_input_channels_);
+  RTC_CHECK_EQ(num_output_channels, num_output_channels_);
+
+  input_buffer_.Write(input, num_input_channels, chunk_size_);
+  size_t first_frame_in_block = frame_offset_;
+
+  // Loop through blocks.
+  while (first_frame_in_block < chunk_size_) {
+    input_buffer_.Read(input_block_.channels(), num_input_channels,
+                       block_size_);
+    input_buffer_.MoveReadPositionBackward(block_size_ - shift_amount_);
+
+    ApplyWindow(window_.get(), block_size_, num_input_channels_,
+                input_block_.channels());
+    callback_->ProcessBlock(input_block_.channels(), block_size_,
+                            num_input_channels_, num_output_channels_,
+                            output_block_.channels());
+    ApplyWindow(window_.get(), block_size_, num_output_channels_,
+                output_block_.channels());
+
+    AddFrames(output_buffer_.channels(), first_frame_in_block,
+              output_block_.channels(), 0, block_size_, num_output_channels_,
+              output_buffer_.channels(), first_frame_in_block);
+
+    first_frame_in_block += shift_amount_;
+  }
+
+  // Copy output buffer to output
+  CopyFrames(output_buffer_.channels(), 0, chunk_size_, num_output_channels_,
+             output, 0);
+
+  // Copy output buffer [chunk_size_, chunk_size_ + initial_delay]
+  // to output buffer [0, initial_delay], zero the rest.
+  MoveFrames(output_buffer_.channels(), chunk_size, initial_delay_,
+             num_output_channels_, output_buffer_.channels(), 0);
+  ZeroOut(output_buffer_.channels(), initial_delay_, chunk_size_,
+          num_output_channels_);
+
+  // Calculate new starting frames.
+  frame_offset_ = first_frame_in_block - chunk_size_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h
new file mode 100644
index 0000000000..59b7e29621
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker.h
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
+
+#include <memory>
+
+#include "common_audio/channel_buffer.h"
+#include "modules/audio_coding/codecs/opus/test/audio_ring_buffer.h"
+
+namespace webrtc {
+
+// The callback function to process audio in the time domain. Input has already
+// been windowed, and output will be windowed. The number of input channels
+// must be >= the number of output channels.
+class BlockerCallback {
+ public:
+  virtual ~BlockerCallback() {}
+
+  virtual void ProcessBlock(const float* const* input,
+                            size_t num_frames,
+                            size_t num_input_channels,
+                            size_t num_output_channels,
+                            float* const* output) = 0;
+};
+
+// The main purpose of Blocker is to abstract away the fact that often we
+// receive a different number of audio frames than our transform takes. For
+// example, most FFTs work best when the fft-size is a power of 2, but suppose
+// we receive 20ms of audio at a sample rate of 48000. That comes to 960 frames
+// of audio, which is not a power of 2. Blocker allows us to specify the
+// transform and all other necessary processing via the Process() callback
+// function without any constraints on the transform-size
+// (read: `block_size_`) or received-audio-size (read: `chunk_size_`).
+// We handle this for the multichannel audio case, allowing for different
+// numbers of input and output channels (for example, beamforming takes 2 or
+// more input channels and returns 1 output channel). Audio signals are
+// represented as deinterleaved floats in the range [-1, 1].
+//
+// Blocker is responsible for:
+// - blocking audio while handling potential discontinuities on the edges
+//   of chunks
+// - windowing blocks before sending them to Process()
+// - windowing processed blocks, and overlap-adding them together before
+//   sending back a processed chunk
+//
+// To use blocker:
+// 1. Impelment a BlockerCallback object `bc`.
+// 2. Instantiate a Blocker object `b`, passing in `bc`.
+// 3. As you receive audio, call b.ProcessChunk() to get processed audio.
+//
+// A small amount of delay is added to the first received chunk to deal with
+// the difference in chunk/block sizes. This delay is <= chunk_size.
+//
+// Ownership of window is retained by the caller.  That is, Blocker makes a
+// copy of window and does not attempt to delete it.
+class Blocker {
+ public:
+  Blocker(size_t chunk_size,
+          size_t block_size,
+          size_t num_input_channels,
+          size_t num_output_channels,
+          const float* window,
+          size_t shift_amount,
+          BlockerCallback* callback);
+  ~Blocker();
+
+  void ProcessChunk(const float* const* input,
+                    size_t chunk_size,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
+                    float* const* output);
+
+  size_t initial_delay() const { return initial_delay_; }
+
+ private:
+  const size_t chunk_size_;
+  const size_t block_size_;
+  const size_t num_input_channels_;
+  const size_t num_output_channels_;
+
+  // The number of frames of delay to add at the beginning of the first chunk.
+  const size_t initial_delay_;
+
+  // The frame index into the input buffer where the first block should be read
+  // from. This is necessary because shift_amount_ is not necessarily a
+  // multiple of chunk_size_, so blocks won't line up at the start of the
+  // buffer.
+  size_t frame_offset_;
+
+  // Since blocks nearly always overlap, there are certain blocks that require
+  // frames from the end of one chunk and the beginning of the next chunk. The
+  // input and output buffers are responsible for saving those frames between
+  // calls to ProcessChunk().
+  //
+  // Both contain |initial delay| + `chunk_size` frames. The input is a fairly
+  // standard FIFO, but due to the overlap-add it's harder to use an
+  // AudioRingBuffer for the output.
+  AudioRingBuffer input_buffer_;
+  ChannelBuffer<float> output_buffer_;
+
+  // Space for the input block (can't wrap because of windowing).
+  ChannelBuffer<float> input_block_;
+
+  // Space for the output block (can't wrap because of overlap/add).
+  ChannelBuffer<float> output_block_;
+
+  std::unique_ptr<float[]> window_;
+
+  // The amount of frames between the start of contiguous blocks. For example,
+  // `shift_amount_` = `block_size_` / 2 for a Hann window.
+  size_t shift_amount_;
+
+  BlockerCallback* callback_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_BLOCKER_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
new file mode 100644
index 0000000000..9c8e789ba9
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/blocker_unittest.cc
@@ -0,0 +1,293 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/blocker.h"
+
+#include <memory>
+
+#include "rtc_base/arraysize.h"
+#include "test/gtest.h"
+
+namespace {
+
+// Callback Function to add 3 to every sample in the signal.
+class PlusThreeBlockerCallback : public webrtc::BlockerCallback {
+ public:
+  void ProcessBlock(const float* const* input,
+                    size_t num_frames,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
+                    float* const* output) override {
+    for (size_t i = 0; i < num_output_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        output[i][j] = input[i][j] + 3;
+      }
+    }
+  }
+};
+
+// No-op Callback Function.
+class CopyBlockerCallback : public webrtc::BlockerCallback {
+ public:
+  void ProcessBlock(const float* const* input,
+                    size_t num_frames,
+                    size_t num_input_channels,
+                    size_t num_output_channels,
+                    float* const* output) override {
+    for (size_t i = 0; i < num_output_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        output[i][j] = input[i][j];
+      }
+    }
+  }
+};
+
+}  // namespace
+
+namespace webrtc {
+
+// Tests blocking with a window that multiplies the signal by 2, a callback
+// that adds 3 to each sample in the signal, and different combinations of chunk
+// size, block size, and shift amount.
+class BlockerTest : public ::testing::Test {
+ protected:
+  void RunTest(Blocker* blocker,
+               size_t chunk_size,
+               size_t num_frames,
+               const float* const* input,
+               float* const* input_chunk,
+               float* const* output,
+               float* const* output_chunk,
+               size_t num_input_channels,
+               size_t num_output_channels) {
+    size_t start = 0;
+    size_t end = chunk_size - 1;
+    while (end < num_frames) {
+      CopyTo(input_chunk, 0, start, num_input_channels, chunk_size, input);
+      blocker->ProcessChunk(input_chunk, chunk_size, num_input_channels,
+                            num_output_channels, output_chunk);
+      CopyTo(output, start, 0, num_output_channels, chunk_size, output_chunk);
+
+      start += chunk_size;
+      end += chunk_size;
+    }
+  }
+
+  void ValidateSignalEquality(const float* const* expected,
+                              const float* const* actual,
+                              size_t num_channels,
+                              size_t num_frames) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        EXPECT_FLOAT_EQ(expected[i][j], actual[i][j]);
+      }
+    }
+  }
+
+  void ValidateInitialDelay(const float* const* output,
+                            size_t num_channels,
+                            size_t num_frames,
+                            size_t initial_delay) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      for (size_t j = 0; j < num_frames; ++j) {
+        if (j < initial_delay) {
+          EXPECT_FLOAT_EQ(output[i][j], 0.f);
+        } else {
+          EXPECT_GT(output[i][j], 0.f);
+        }
+      }
+    }
+  }
+
+  static void CopyTo(float* const* dst,
+                     size_t start_index_dst,
+                     size_t start_index_src,
+                     size_t num_channels,
+                     size_t num_frames,
+                     const float* const* src) {
+    for (size_t i = 0; i < num_channels; ++i) {
+      memcpy(&dst[i][start_index_dst], &src[i][start_index_src],
+             num_frames * sizeof(float));
+    }
+  }
+};
+
+TEST_F(BlockerTest, TestBlockerMutuallyPrimeChunkandBlockSize) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 10;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 5;
+  const size_t kShiftAmount = 2;
+
+  const float kInput[kNumInputChannels][kNumFrames] = {
+      {1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
+      {2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
+      {3, 3, 3, 3, 3, 3, 3, 3, 3, 3}};
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput));
+
+  const float kExpectedOutput[kNumInputChannels][kNumFrames] = {
+      {6, 6, 12, 20, 20, 20, 20, 20, 20, 20},
+      {6, 6, 12, 28, 28, 28, 28, 28, 28, 28}};
+  ChannelBuffer<float> expected_output_cb(kNumFrames, kNumInputChannels);
+  expected_output_cb.SetDataForTesting(
+      kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput));
+
+  const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f};
+
+  ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels);
+  ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels);
+  ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
+
+  PlusThreeBlockerCallback callback;
+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
+                  kWindow, kShiftAmount, &callback);
+
+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
+          input_chunk_cb.channels(), actual_output_cb.channels(),
+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+  ValidateSignalEquality(expected_output_cb.channels(),
+                         actual_output_cb.channels(), kNumOutputChannels,
+                         kNumFrames);
+}
+
+TEST_F(BlockerTest, TestBlockerMutuallyPrimeShiftAndBlockSize) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 12;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 6;
+  const size_t kShiftAmount = 3;
+
+  const float kInput[kNumInputChannels][kNumFrames] = {
+      {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
+      {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
+      {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}};
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput));
+
+  const float kExpectedOutput[kNumOutputChannels][kNumFrames] = {
+      {6, 10, 10, 20, 10, 10, 20, 10, 10, 20, 10, 10},
+      {6, 14, 14, 28, 14, 14, 28, 14, 14, 28, 14, 14}};
+  ChannelBuffer<float> expected_output_cb(kNumFrames, kNumOutputChannels);
+  expected_output_cb.SetDataForTesting(
+      kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput));
+
+  const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f};
+
+  ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels);
+  ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels);
+  ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
+
+  PlusThreeBlockerCallback callback;
+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
+                  kWindow, kShiftAmount, &callback);
+
+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
+          input_chunk_cb.channels(), actual_output_cb.channels(),
+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+  ValidateSignalEquality(expected_output_cb.channels(),
+                         actual_output_cb.channels(), kNumOutputChannels,
+                         kNumFrames);
+}
+
+TEST_F(BlockerTest, TestBlockerNoOverlap) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 12;
+  const size_t kBlockSize = 4;
+  const size_t kChunkSize = 4;
+  const size_t kShiftAmount = 4;
+
+  const float kInput[kNumInputChannels][kNumFrames] = {
+      {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
+      {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
+      {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3}};
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(kInput[0], sizeof(kInput) / sizeof(**kInput));
+
+  const float kExpectedOutput[kNumOutputChannels][kNumFrames] = {
+      {10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10},
+      {14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14, 14}};
+  ChannelBuffer<float> expected_output_cb(kNumFrames, kNumOutputChannels);
+  expected_output_cb.SetDataForTesting(
+      kExpectedOutput[0], sizeof(kExpectedOutput) / sizeof(**kExpectedOutput));
+
+  const float kWindow[kBlockSize] = {2.f, 2.f, 2.f, 2.f};
+
+  ChannelBuffer<float> actual_output_cb(kNumFrames, kNumOutputChannels);
+  ChannelBuffer<float> input_chunk_cb(kChunkSize, kNumInputChannels);
+  ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
+
+  PlusThreeBlockerCallback callback;
+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
+                  kWindow, kShiftAmount, &callback);
+
+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
+          input_chunk_cb.channels(), actual_output_cb.channels(),
+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+  ValidateSignalEquality(expected_output_cb.channels(),
+                         actual_output_cb.channels(), kNumOutputChannels,
+                         kNumFrames);
+}
+
+TEST_F(BlockerTest, InitialDelaysAreMinimum) {
+  const size_t kNumInputChannels = 3;
+  const size_t kNumOutputChannels = 2;
+  const size_t kNumFrames = 1280;
+  const size_t kChunkSize[] = {80,  80,  80,  80,  80,  80,
+                               160, 160, 160, 160, 160, 160};
+  const size_t kBlockSize[] = {64,  64,  64,  128, 128, 128,
+                               128, 128, 128, 256, 256, 256};
+  const size_t kShiftAmount[] = {16, 32, 64,  32, 64,  128,
+                                 32, 64, 128, 64, 128, 256};
+  const size_t kInitialDelay[] = {48, 48, 48, 112, 112, 112,
+                                  96, 96, 96, 224, 224, 224};
+
+  float input[kNumInputChannels][kNumFrames];
+  for (size_t i = 0; i < kNumInputChannels; ++i) {
+    for (size_t j = 0; j < kNumFrames; ++j) {
+      input[i][j] = i + 1;
+    }
+  }
+  ChannelBuffer<float> input_cb(kNumFrames, kNumInputChannels);
+  input_cb.SetDataForTesting(input[0], sizeof(input) / sizeof(**input));
+
+  ChannelBuffer<float> output_cb(kNumFrames, kNumOutputChannels);
+
+  CopyBlockerCallback callback;
+
+  for (size_t i = 0; i < arraysize(kChunkSize); ++i) {
+    std::unique_ptr<float[]> window(new float[kBlockSize[i]]);
+    for (size_t j = 0; j < kBlockSize[i]; ++j) {
+      window[j] = 1.f;
+    }
+
+    ChannelBuffer<float> input_chunk_cb(kChunkSize[i], kNumInputChannels);
+    ChannelBuffer<float> output_chunk_cb(kChunkSize[i], kNumOutputChannels);
+
+    Blocker blocker(kChunkSize[i], kBlockSize[i], kNumInputChannels,
+                    kNumOutputChannels, window.get(), kShiftAmount[i],
+                    &callback);
+
+    RunTest(&blocker, kChunkSize[i], kNumFrames, input_cb.channels(),
+            input_chunk_cb.channels(), output_cb.channels(),
+            output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
+
+    ValidateInitialDelay(output_cb.channels(), kNumOutputChannels, kNumFrames,
+                         kInitialDelay[i]);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc
new file mode 100644
index 0000000000..b1a6526bba
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.cc
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
+
+#include <algorithm>
+#include <cstdlib>
+#include <cstring>
+
+#include "common_audio/real_fourier.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void LappedTransform::BlockThunk::ProcessBlock(const float* const* input,
+                                               size_t num_frames,
+                                               size_t num_input_channels,
+                                               size_t num_output_channels,
+                                               float* const* output) {
+  RTC_CHECK_EQ(num_input_channels, parent_->num_in_channels_);
+  RTC_CHECK_EQ(num_output_channels, parent_->num_out_channels_);
+  RTC_CHECK_EQ(parent_->block_length_, num_frames);
+
+  for (size_t i = 0; i < num_input_channels; ++i) {
+    memcpy(parent_->real_buf_.Row(i), input[i], num_frames * sizeof(*input[0]));
+    parent_->fft_->Forward(parent_->real_buf_.Row(i),
+                           parent_->cplx_pre_.Row(i));
+  }
+
+  size_t block_length =
+      RealFourier::ComplexLength(RealFourier::FftOrder(num_frames));
+  RTC_CHECK_EQ(parent_->cplx_length_, block_length);
+  parent_->block_processor_->ProcessAudioBlock(
+      parent_->cplx_pre_.Array(), num_input_channels, parent_->cplx_length_,
+      num_output_channels, parent_->cplx_post_.Array());
+
+  for (size_t i = 0; i < num_output_channels; ++i) {
+    parent_->fft_->Inverse(parent_->cplx_post_.Row(i),
+                           parent_->real_buf_.Row(i));
+    memcpy(output[i], parent_->real_buf_.Row(i),
+           num_frames * sizeof(*input[0]));
+  }
+}
+
+LappedTransform::LappedTransform(size_t num_in_channels,
+                                 size_t num_out_channels,
+                                 size_t chunk_length,
+                                 const float* window,
+                                 size_t block_length,
+                                 size_t shift_amount,
+                                 Callback* callback)
+    : blocker_callback_(this),
+      num_in_channels_(num_in_channels),
+      num_out_channels_(num_out_channels),
+      block_length_(block_length),
+      chunk_length_(chunk_length),
+      block_processor_(callback),
+      blocker_(chunk_length_,
+               block_length_,
+               num_in_channels_,
+               num_out_channels_,
+               window,
+               shift_amount,
+               &blocker_callback_),
+      fft_(RealFourier::Create(RealFourier::FftOrder(block_length_))),
+      cplx_length_(RealFourier::ComplexLength(fft_->order())),
+      real_buf_(num_in_channels,
+                block_length_,
+                RealFourier::kFftBufferAlignment),
+      cplx_pre_(num_in_channels,
+                cplx_length_,
+                RealFourier::kFftBufferAlignment),
+      cplx_post_(num_out_channels,
+                 cplx_length_,
+                 RealFourier::kFftBufferAlignment) {
+  RTC_CHECK(num_in_channels_ > 0);
+  RTC_CHECK_GT(block_length_, 0);
+  RTC_CHECK_GT(chunk_length_, 0);
+  RTC_CHECK(block_processor_);
+
+  // block_length_ power of 2?
+  RTC_CHECK_EQ(0, block_length_ & (block_length_ - 1));
+}
+
+LappedTransform::~LappedTransform() = default;
+
+void LappedTransform::ProcessChunk(const float* const* in_chunk,
+                                   float* const* out_chunk) {
+  blocker_.ProcessChunk(in_chunk, chunk_length_, num_in_channels_,
+                        num_out_channels_, out_chunk);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h
new file mode 100644
index 0000000000..bb25c34a9e
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform.h
@@ -0,0 +1,175 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
+
+#include <complex>
+#include <memory>
+
+#include "common_audio/real_fourier.h"
+#include "modules/audio_coding/codecs/opus/test/blocker.h"
+#include "rtc_base/memory/aligned_malloc.h"
+
+namespace webrtc {
+
+// Wrapper class for aligned arrays. Every row (and the first dimension) are
+// aligned to the given byte alignment.
+template <typename T>
+class AlignedArray {
+ public:
+  AlignedArray(size_t rows, size_t cols, size_t alignment)
+      : rows_(rows), cols_(cols) {
+    RTC_CHECK_GT(alignment, 0);
+    head_row_ =
+        static_cast<T**>(AlignedMalloc(rows_ * sizeof(*head_row_), alignment));
+    for (size_t i = 0; i < rows_; ++i) {
+      head_row_[i] = static_cast<T*>(
+          AlignedMalloc(cols_ * sizeof(**head_row_), alignment));
+    }
+  }
+
+  ~AlignedArray() {
+    for (size_t i = 0; i < rows_; ++i) {
+      AlignedFree(head_row_[i]);
+    }
+    AlignedFree(head_row_);
+  }
+
+  T* const* Array() { return head_row_; }
+
+  const T* const* Array() const { return head_row_; }
+
+  T* Row(size_t row) {
+    RTC_CHECK_LE(row, rows_);
+    return head_row_[row];
+  }
+
+  const T* Row(size_t row) const {
+    RTC_CHECK_LE(row, rows_);
+    return head_row_[row];
+  }
+
+ private:
+  size_t rows_;
+  size_t cols_;
+  T** head_row_;
+};
+
+// Helper class for audio processing modules which operate on frequency domain
+// input derived from the windowed time domain audio stream.
+//
+// The input audio chunk is sliced into possibly overlapping blocks, multiplied
+// by a window and transformed with an FFT implementation. The transformed data
+// is supplied to the given callback for processing. The processed output is
+// then inverse transformed into the time domain and spliced back into a chunk
+// which constitutes the final output of this processing module.
+class LappedTransform {
+ public:
+  class Callback {
+   public:
+    virtual ~Callback() {}
+
+    virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
+                                   size_t num_in_channels,
+                                   size_t frames,
+                                   size_t num_out_channels,
+                                   std::complex<float>* const* out_block) = 0;
+  };
+
+  // Construct a transform instance. `chunk_length` is the number of samples in
+  // each channel. `window` defines the window, owned by the caller (a copy is
+  // made internally); `window` should have length equal to `block_length`.
+  // `block_length` defines the length of a block, in samples.
+  // `shift_amount` is in samples. `callback` is the caller-owned audio
+  // processing function called for each block of the input chunk.
+  LappedTransform(size_t num_in_channels,
+                  size_t num_out_channels,
+                  size_t chunk_length,
+                  const float* window,
+                  size_t block_length,
+                  size_t shift_amount,
+                  Callback* callback);
+  ~LappedTransform();
+
+  // Main audio processing helper method. Internally slices `in_chunk` into
+  // blocks, transforms them to frequency domain, calls the callback for each
+  // block and returns a de-blocked time domain chunk of audio through
+  // `out_chunk`. Both buffers are caller-owned.
+  void ProcessChunk(const float* const* in_chunk, float* const* out_chunk);
+
+  // Get the chunk length.
+  //
+  // The chunk length is the number of samples per channel that must be passed
+  // to ProcessChunk via the parameter in_chunk.
+  //
+  // Returns the same chunk_length passed to the LappedTransform constructor.
+  size_t chunk_length() const { return chunk_length_; }
+
+  // Get the number of input channels.
+  //
+  // This is the number of arrays that must be passed to ProcessChunk via
+  // in_chunk.
+  //
+  // Returns the same num_in_channels passed to the LappedTransform constructor.
+  size_t num_in_channels() const { return num_in_channels_; }
+
+  // Get the number of output channels.
+  //
+  // This is the number of arrays that must be passed to ProcessChunk via
+  // out_chunk.
+  //
+  // Returns the same num_out_channels passed to the LappedTransform
+  // constructor.
+  size_t num_out_channels() const { return num_out_channels_; }
+
+  // Returns the initial delay.
+  //
+  // This is the delay introduced by the `blocker_` to be able to get and return
+  // chunks of `chunk_length`, but process blocks of `block_length`.
+  size_t initial_delay() const { return blocker_.initial_delay(); }
+
+ private:
+  // Internal middleware callback, given to the blocker. Transforms each block
+  // and hands it over to the processing method given at construction time.
+  class BlockThunk : public BlockerCallback {
+   public:
+    explicit BlockThunk(LappedTransform* parent) : parent_(parent) {}
+
+    void ProcessBlock(const float* const* input,
+                      size_t num_frames,
+                      size_t num_input_channels,
+                      size_t num_output_channels,
+                      float* const* output) override;
+
+   private:
+    LappedTransform* const parent_;
+  } blocker_callback_;
+
+  const size_t num_in_channels_;
+  const size_t num_out_channels_;
+
+  const size_t block_length_;
+  const size_t chunk_length_;
+
+  Callback* const block_processor_;
+  Blocker blocker_;
+
+  // TODO(alessiob): Replace RealFourier with a different FFT library.
+  std::unique_ptr<RealFourier> fft_;
+  const size_t cplx_length_;
+  AlignedArray<float> real_buf_;
+  AlignedArray<std::complex<float> > cplx_pre_;
+  AlignedArray<std::complex<float> > cplx_post_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_CODING_CODECS_OPUS_TEST_LAPPED_TRANSFORM_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc
new file mode 100644
index 0000000000..1003ed52e5
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/test/lapped_transform_unittest.cc
@@ -0,0 +1,203 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstring>
+
+#include "test/gtest.h"
+
+using std::complex;
+
+namespace {
+
+class NoopCallback : public webrtc::LappedTransform::Callback {
+ public:
+  NoopCallback() : block_num_(0) {}
+
+  void ProcessAudioBlock(const complex<float>* const* in_block,
+                         size_t in_channels,
+                         size_t frames,
+                         size_t out_channels,
+                         complex<float>* const* out_block) override {
+    RTC_CHECK_EQ(in_channels, out_channels);
+    for (size_t i = 0; i < out_channels; ++i) {
+      memcpy(out_block[i], in_block[i], sizeof(**in_block) * frames);
+    }
+    ++block_num_;
+  }
+
+  size_t block_num() { return block_num_; }
+
+ private:
+  size_t block_num_;
+};
+
+class FftCheckerCallback : public webrtc::LappedTransform::Callback {
+ public:
+  FftCheckerCallback() : block_num_(0) {}
+
+  void ProcessAudioBlock(const complex<float>* const* in_block,
+                         size_t in_channels,
+                         size_t frames,
+                         size_t out_channels,
+                         complex<float>* const* out_block) override {
+    RTC_CHECK_EQ(in_channels, out_channels);
+
+    size_t full_length = (frames - 1) * 2;
+    ++block_num_;
+
+    if (block_num_ > 0) {
+      ASSERT_NEAR(in_block[0][0].real(), static_cast<float>(full_length),
+                  1e-5f);
+      ASSERT_NEAR(in_block[0][0].imag(), 0.0f, 1e-5f);
+      for (size_t i = 1; i < frames; ++i) {
+        ASSERT_NEAR(in_block[0][i].real(), 0.0f, 1e-5f);
+        ASSERT_NEAR(in_block[0][i].imag(), 0.0f, 1e-5f);
+      }
+    }
+  }
+
+  size_t block_num() { return block_num_; }
+
+ private:
+  size_t block_num_;
+};
+
+void SetFloatArray(float value, int rows, int cols, float* const* array) {
+  for (int i = 0; i < rows; ++i) {
+    for (int j = 0; j < cols; ++j) {
+      array[i][j] = value;
+    }
+  }
+}
+
+}  // namespace
+
+namespace webrtc {
+
+TEST(LappedTransformTest, Windowless) {
+  const size_t kChannels = 3;
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  const size_t kShiftAmount = 64;
+  NoopCallback noop;
+
+  // Rectangular window.
+  float window[kBlockLength];
+  std::fill(window, &window[kBlockLength], 1.0f);
+
+  LappedTransform trans(kChannels, kChannels, kChunkLength, window,
+                        kBlockLength, kShiftAmount, &noop);
+  float in_buffer[kChannels][kChunkLength];
+  float* in_chunk[kChannels];
+  float out_buffer[kChannels][kChunkLength];
+  float* out_chunk[kChannels];
+
+  in_chunk[0] = in_buffer[0];
+  in_chunk[1] = in_buffer[1];
+  in_chunk[2] = in_buffer[2];
+  out_chunk[0] = out_buffer[0];
+  out_chunk[1] = out_buffer[1];
+  out_chunk[2] = out_buffer[2];
+  SetFloatArray(2.0f, kChannels, kChunkLength, in_chunk);
+  SetFloatArray(-1.0f, kChannels, kChunkLength, out_chunk);
+
+  trans.ProcessChunk(in_chunk, out_chunk);
+
+  for (size_t i = 0; i < kChannels; ++i) {
+    for (size_t j = 0; j < kChunkLength; ++j) {
+      ASSERT_NEAR(out_chunk[i][j], 2.0f, 1e-5f);
+    }
+  }
+
+  ASSERT_EQ(kChunkLength / kBlockLength, noop.block_num());
+}
+
+TEST(LappedTransformTest, IdentityProcessor) {
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  const size_t kShiftAmount = 32;
+  NoopCallback noop;
+
+  // Identity window for |overlap = block_size / 2|.
+  float window[kBlockLength];
+  std::fill(window, &window[kBlockLength], std::sqrt(0.5f));
+
+  LappedTransform trans(1, 1, kChunkLength, window, kBlockLength, kShiftAmount,
+                        &noop);
+  float in_buffer[kChunkLength];
+  float* in_chunk = in_buffer;
+  float out_buffer[kChunkLength];
+  float* out_chunk = out_buffer;
+
+  SetFloatArray(2.0f, 1, kChunkLength, &in_chunk);
+  SetFloatArray(-1.0f, 1, kChunkLength, &out_chunk);
+
+  trans.ProcessChunk(&in_chunk, &out_chunk);
+
+  for (size_t i = 0; i < kChunkLength; ++i) {
+    ASSERT_NEAR(out_chunk[i], (i < kBlockLength - kShiftAmount) ? 0.0f : 2.0f,
+                1e-5f);
+  }
+
+  ASSERT_EQ(kChunkLength / kShiftAmount, noop.block_num());
+}
+
+TEST(LappedTransformTest, Callbacks) {
+  const size_t kChunkLength = 512;
+  const size_t kBlockLength = 64;
+  FftCheckerCallback call;
+
+  // Rectangular window.
+  float window[kBlockLength];
+  std::fill(window, &window[kBlockLength], 1.0f);
+
+  LappedTransform trans(1, 1, kChunkLength, window, kBlockLength, kBlockLength,
+                        &call);
+  float in_buffer[kChunkLength];
+  float* in_chunk = in_buffer;
+  float out_buffer[kChunkLength];
+  float* out_chunk = out_buffer;
+
+  SetFloatArray(1.0f, 1, kChunkLength, &in_chunk);
+  SetFloatArray(-1.0f, 1, kChunkLength, &out_chunk);
+
+  trans.ProcessChunk(&in_chunk, &out_chunk);
+
+  ASSERT_EQ(kChunkLength / kBlockLength, call.block_num());
+}
+
+TEST(LappedTransformTest, chunk_length) {
+  const size_t kBlockLength = 64;
+  FftCheckerCallback call;
+  const float window[kBlockLength] = {};
+
+  // Make sure that chunk_length returns the same value passed to the
+  // LappedTransform constructor.
+  {
+    const size_t kExpectedChunkLength = 512;
+    const LappedTransform trans(1, 1, kExpectedChunkLength, window,
+                                kBlockLength, kBlockLength, &call);
+
+    EXPECT_EQ(kExpectedChunkLength, trans.chunk_length());
+  }
+  {
+    const size_t kExpectedChunkLength = 160;
+    const LappedTransform trans(1, 1, kExpectedChunkLength, window,
+                                kBlockLength, kBlockLength, &call);
+
+    EXPECT_EQ(kExpectedChunkLength, trans.chunk_length());
+  }
+}
+
+}  // namespace webrtc