text: /*
   *  Copyright (c) 2008 The WebRTC project authors. All Rights Reserved.
   *
   *  Use of this source code is governed by a BSD-style license
   *  that can be found in the LICENSE file in the root of the source
   *  tree. An additional intellectual property rights grant can 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 <utility>
  
  #include "absl/memory/memory.h"
  #include "absl/strings/match.h"
  #include "api/audio_codecs/builtin_audio_decoder_factory.h"
  #include "api/audio_codecs/builtin_audio_encoder_factory.h"
  #include "api/media_transport_config.h"
  #include "api/rtp_parameters.h"
  #include "api/scoped_refptr.h"
  #include "api/task_queue/default_task_queue_factory.h"
  #include "call/call.h"
  #include "logging/rtc_event_log/rtc_event_log.h"
  #include "media/base/fake_media_engine.h"
  #include "media/base/fake_network_interface.h"
  #include "media/base/fake_rtp.h"
  #include "media/base/media_constants.h"
  #include "media/engine/fake_webrtc_call.h"
  #include "media/engine/webrtc_voice_engine.h"
  #include "modules/audio_device/include/mock_audio_device.h"
  #include "modules/audio_processing/include/mock_audio_processing.h"
  #include "rtc_base/arraysize.h"
  #include "rtc_base/byte_order.h"
  #include "rtc_base/numerics/safe_conversions.h"
  #include "test/field_trial.h"
  #include "test/gtest.h"
  #include "test/mock_audio_decoder_factory.h"
  #include "test/mock_audio_encoder_factory.h"
  
  using ::testing::_;
  using ::testing::ContainerEq;
  using ::testing::Contains;
  using ::testing::Field;
  using ::testing::Return;
  using ::testing::ReturnPointee;
  using ::testing::SaveArg;
  using ::testing::StrictMock;
  
  namespace {
  using webrtc::BitrateConstraints;
  
  constexpr uint32_t kMaxUnsignaledRecvStreams = 4;
  
  const cricket::AudioCodec kPcmuCodec(0, "PCMU", 8000, 64000, 1);
  const cricket::AudioCodec kIsacCodec(103, "ISAC", 16000, 32000, 1);
  const cricket::AudioCodec kOpusCodec(111, "opus", 48000, 32000, 2);
  const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1);
  const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1);
  const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1);
  const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1);
  const cricket::AudioCodec kTelephoneEventCodec1(106,
                                                  "telephone-event",
                                                  8000,
                                                  0,
                                                  1);
  const cricket::AudioCodec kTelephoneEventCodec2(107,
                                                  "telephone-event",
                                                  32000,
                                                  0,
                                                  1);
  
  const uint32_t kSsrc0 = 0;
  const uint32_t kSsrc1 = 1;
  const uint32_t kSsrcX = 0x99;
  const uint32_t kSsrcY = 0x17;
  const uint32_t kSsrcZ = 0x42;
  const uint32_t kSsrcW = 0x02;
  const uint32_t kSsrcs4[] = {11, 200, 30, 44};
  
  constexpr int kRtpHistoryMs = 5000;
  
  constexpr webrtc::AudioProcessing::Config::GainController1::Mode
      kDefaultAgcMode =
  #if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
          webrtc::AudioProcessing::Config::GainController1::kFixedDigital;
  #else
          webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
  #endif
  
  constexpr webrtc::NoiseSuppression::Level kDefaultNsLevel =
      webrtc::NoiseSuppression::kHigh;
  
  void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
    RTC_DCHECK(adm);
  
    // Setup.
    EXPECT_CALL(*adm, AddRef()).Times(3);
    EXPECT_CALL(*adm, Init()).WillOnce(Return(0));
    EXPECT_CALL(*adm, RegisterAudioCallback(_)).WillOnce(Return(0));
  #if defined(WEBRTC_WIN)
    EXPECT_CALL(
        *adm,
        SetPlayoutDevice(
            ::testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
                webrtc::AudioDeviceModule::kDefaultCommunicationDevice)))
        .WillOnce(Return(0));
  #else
    EXPECT_CALL(*adm, SetPlayoutDevice(0)).WillOnce(Return(0));
  #endif  // #if defined(WEBRTC_WIN)
    EXPECT_CALL(*adm, InitSpeaker()).WillOnce(Return(0));
    EXPECT_CALL(*adm, StereoPlayoutIsAvailable(::testing::_)).WillOnce(Return(0));
    EXPECT_CALL(*adm, SetStereoPlayout(false)).WillOnce(Return(0));
  #if defined(WEBRTC_WIN)
    EXPECT_CALL(
        *adm,
        SetRecordingDevice(
            ::testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
                webrtc::AudioDeviceModule::kDefaultCommunicationDevice)))
        .WillOnce(Return(0));
  #else
    EXPECT_CALL(*adm, SetRecordingDevice(0)).WillOnce(Return(0));
  #endif  // #if defined(WEBRTC_WIN)
    EXPECT_CALL(*adm, InitMicrophone()).WillOnce(Return(0));
    EXPECT_CALL(*adm, StereoRecordingIsAvailable(::testing::_))
        .WillOnce(Return(0));
    EXPECT_CALL(*adm, SetStereoRecording(false)).WillOnce(Return(0));
    EXPECT_CALL(*adm, BuiltInAECIsAvailable()).WillOnce(Return(false));
    EXPECT_CALL(*adm, BuiltInAGCIsAvailable()).WillOnce(Return(false));
    EXPECT_CALL(*adm, BuiltInNSIsAvailable()).WillOnce(Return(false));
  
    // Teardown.
    EXPECT_CALL(*adm, StopPlayout()).WillOnce(Return(0));
    EXPECT_CALL(*adm, StopRecording()).WillOnce(Return(0));
    EXPECT_CALL(*adm, RegisterAudioCallback(nullptr)).WillOnce(Return(0));
    EXPECT_CALL(*adm, Terminate()).WillOnce(Return(0));
    EXPECT_CALL(*adm, Release())
        .Times(3)
        .WillRepeatedly(Return(rtc::RefCountReleaseStatus::kDroppedLastRef));
  }
  }  // namespace
  
  // Tests that our stub library "works".
  TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
        webrtc::CreateDefaultTaskQueueFactory();
    StrictMock<webrtc::test::MockAudioDeviceModule> adm;
    AdmSetupExpectations(&adm);
    rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm =
        new rtc::RefCountedObject<
            StrictMock<webrtc::test::MockAudioProcessing>>();
    webrtc::AudioProcessing::Config apm_config;
    EXPECT_CALL(*apm, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config));
    EXPECT_CALL(*apm, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config));
    EXPECT_CALL(*apm, SetExtraOptions(::testing::_));
    EXPECT_CALL(*apm, DetachAecDump());
    {
      cricket::WebRtcVoiceEngine engine(
          task_queue_factory.get(), &adm,
          webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
          webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
      engine.Init();
    }
  }
  
  class FakeAudioSink : public webrtc::AudioSinkInterface {
   public:
    void OnData(const Data& audio) override {}
  };
  
  class FakeAudioSource : public cricket::AudioSource {
    void SetSink(Sink* sink) override {}
  };
  
  class WebRtcVoiceEngineTestFake : public ::testing::Test {
   public:
    WebRtcVoiceEngineTestFake() : WebRtcVoiceEngineTestFake("") {}
  
    explicit WebRtcVoiceEngineTestFake(const char* field_trials)
        : task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
          apm_(new rtc::RefCountedObject<
               StrictMock<webrtc::test::MockAudioProcessing>>()),
          apm_ns_(*apm_->noise_suppression()),
          call_(),
          override_field_trials_(field_trials) {
      // AudioDeviceModule.
      AdmSetupExpectations(&adm_);
      // AudioProcessing.
      EXPECT_CALL(*apm_, GetConfig()).WillRepeatedly(ReturnPointee(&apm_config_));
      EXPECT_CALL(*apm_, ApplyConfig(_)).WillRepeatedly(SaveArg<0>(&apm_config_));
      EXPECT_CALL(*apm_, SetExtraOptions(::testing::_));
      EXPECT_CALL(*apm_, DetachAecDump());
      // Default Options.
      EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
      EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0));
      // TODO(kwiberg): We should use mock factories here, but a bunch of
      // the tests here probe the specific set of codecs provided by the builtin
      // factories. Those tests should probably be moved elsewhere.
      auto encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
      auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
      engine_.reset(new cricket::WebRtcVoiceEngine(
          task_queue_factory_.get(), &adm_, encoder_factory, decoder_factory,
          nullptr, apm_));
      engine_->Init();
      send_parameters_.codecs.push_back(kPcmuCodec);
      recv_parameters_.codecs.push_back(kPcmuCodec);
  
      // Default Options.
      EXPECT_TRUE(IsEchoCancellationEnabled());
      EXPECT_TRUE(IsHighPassFilterEnabled());
      EXPECT_TRUE(IsTypingDetectionEnabled());
      VerifyGainControlEnabledCorrectly();
      VerifyGainControlDefaultSettings();
    }
  
    bool SetupChannel() {
      EXPECT_CALL(*apm_, SetExtraOptions(::testing::_));
      channel_ = engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
                                             cricket::AudioOptions(),
                                             webrtc::CryptoOptions());
      return (channel_ != nullptr);
    }
  
    bool SetupRecvStream() {
      if (!SetupChannel()) {
        return false;
      }
      return AddRecvStream(kSsrcX);
    }
  
    bool SetupSendStream() {
      return SetupSendStream(cricket::StreamParams::CreateLegacy(kSsrcX));
    }
  
    bool SetupSendStream(const cricket::StreamParams& sp) {
      if (!SetupChannel()) {
        return false;
      }
      if (!channel_->AddSendStream(sp)) {
        return false;
      }
      EXPECT_CALL(*apm_, set_output_will_be_muted(false));
      return channel_->SetAudioSend(kSsrcX, true, nullptr, &fake_source_);
    }
  
    bool AddRecvStream(uint32_t ssrc) {
      EXPECT_TRUE(channel_);
      return channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(ssrc));
    }
  
    void SetupForMultiSendStream() {
      EXPECT_TRUE(SetupSendStream());
      // Remove stream added in Setup.
      EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
      EXPECT_TRUE(channel_->RemoveSendStream(kSsrcX));
      // Verify the channel does not exist.
      EXPECT_FALSE(call_.GetAudioSendStream(kSsrcX));
    }
  
    void DeliverPacket(const void* data, int len) {
      rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len);
      channel_->OnPacketReceived(packet, /* packet_time_us */ -1);
    }
  
    void TearDown() override { delete channel_; }
  
    const cricket::FakeAudioSendStream& GetSendStream(uint32_t ssrc) {
      const auto* send_stream = call_.GetAudioSendStream(ssrc);
      EXPECT_TRUE(send_stream);
      return *send_stream;
    }
  
    const cricket::FakeAudioReceiveStream& GetRecvStream(uint32_t ssrc) {
      const auto* recv_stream = call_.GetAudioReceiveStream(ssrc);
      EXPECT_TRUE(recv_stream);
      return *recv_stream;
    }
  
    const webrtc::AudioSendStream::Config& GetSendStreamConfig(uint32_t ssrc) {
      return GetSendStream(ssrc).GetConfig();
    }
  
    const webrtc::AudioReceiveStream::Config& GetRecvStreamConfig(uint32_t ssrc) {
      return GetRecvStream(ssrc).GetConfig();
    }
  
    void SetSend(bool enable) {
      ASSERT_TRUE(channel_);
      if (enable) {
        EXPECT_CALL(adm_, RecordingIsInitialized()).WillOnce(Return(false));
        EXPECT_CALL(adm_, Recording()).WillOnce(Return(false));
        EXPECT_CALL(adm_, InitRecording()).WillOnce(Return(0));
        EXPECT_CALL(*apm_, SetExtraOptions(::testing::_));
      }
      channel_->SetSend(enable);
    }
  
    void SetSendParameters(const cricket::AudioSendParameters& params) {
      EXPECT_CALL(*apm_, SetExtraOptions(::testing::_));
      ASSERT_TRUE(channel_);
      EXPECT_TRUE(channel_->SetSendParameters(params));
    }
  
    void SetAudioSend(uint32_t ssrc,
                      bool enable,
                      cricket::AudioSource* source,
                      const cricket::AudioOptions* options = nullptr) {
      EXPECT_CALL(*apm_, set_output_will_be_muted(!enable));
      ASSERT_TRUE(channel_);
      if (enable && options) {
        EXPECT_CALL(*apm_, SetExtraOptions(::testing::_));
      }
      EXPECT_TRUE(channel_->SetAudioSend(ssrc, enable, options, source));
    }
  
    void TestInsertDtmf(uint32_t ssrc,
                        bool caller,
                        const cricket::AudioCodec& codec) {
      EXPECT_TRUE(SetupChannel());
      if (caller) {
        // If this is a caller, local description will be applied and add the
        // send stream.
        EXPECT_TRUE(
            channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
      }
  
      // Test we can only InsertDtmf when the other side supports telephone-event.
      SetSendParameters(send_parameters_);
      SetSend(true);
      EXPECT_FALSE(channel_->CanInsertDtmf());
      EXPECT_FALSE(channel_->InsertDtmf(ssrc, 1, 111));
      send_parameters_.codecs.push_back(codec);
      SetSendParameters(send_parameters_);
      EXPECT_TRUE(channel_->CanInsertDtmf());
  
      if (!caller) {
        // If this is callee, there's no active send channel yet.
        EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123));
        EXPECT_TRUE(
            channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
      }
  
      // Check we fail if the ssrc is invalid.
      EXPECT_FALSE(channel_->InsertDtmf(-1, 1, 111));
  
      // Test send.
      cricket::FakeAudioSendStream::TelephoneEvent telephone_event =
          GetSendStream(kSsrcX).GetLatestTelephoneEvent();
      EXPECT_EQ(-1, telephone_event.payload_type);
      EXPECT_TRUE(channel_->InsertDtmf(ssrc, 2, 123));
      telephone_event = GetSendStream(kSsrcX).GetLatestTelephoneEvent();
      EXPECT_EQ(codec.id, telephone_event.payload_type);
      EXPECT_EQ(codec.clockrate, telephone_event.payload_frequency);
      EXPECT_EQ(2, telephone_event.event_code);
      EXPECT_EQ(123, telephone_event.duration_ms);
    }
  
    void TestExtmapAllowMixedCaller(bool extmap_allow_mixed) {
      // For a caller, the answer will be applied in set remote description
      // where SetSendParameters() is called.
      EXPECT_TRUE(SetupChannel());
      EXPECT_TRUE(
          channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
      send_parameters_.extmap_allow_mixed = extmap_allow_mixed;
      SetSendParameters(send_parameters_);
      const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX);
      EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
    }
  
    void TestExtmapAllowMixedCallee(bool extmap_allow_mixed) {
      // For a callee, the answer will be applied in set local description
      // where SetExtmapAllowMixed() and AddSendStream() are called.
      EXPECT_TRUE(SetupChannel());
      channel_->SetExtmapAllowMixed(extmap_allow_mixed);
      EXPECT_TRUE(
          channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
  
      const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX);
      EXPECT_EQ(extmap_allow_mixed, config.rtp.extmap_allow_mixed);
    }
  
    // Test that send bandwidth is set correctly.
    // |codec| is the codec under test.
    // |max_bitrate| is a parameter to set to SetMaxSendBandwidth().
    // |expected_result| is the expected result from SetMaxSendBandwidth().
    // |expected_bitrate| is the expected audio bitrate afterward.
    void TestMaxSendBandwidth(const cricket::AudioCodec& codec,
                              int max_bitrate,
                              bool expected_result,
                              int expected_bitrate) {
      cricket::AudioSendParameters parameters;
      parameters.codecs.push_back(codec);
      parameters.max_bandwidth_bps = max_bitrate;
      if (expected_result) {
        SetSendParameters(parameters);
      } else {
        EXPECT_FALSE(channel_->SetSendParameters(parameters));
      }
      EXPECT_EQ(expected_bitrate, GetCodecBitrate(kSsrcX));
    }
  
    // Sets the per-stream maximum bitrate limit for the specified SSRC.
    bool SetMaxBitrateForStream(int32_t ssrc, int bitrate) {
      webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(ssrc);
      EXPECT_EQ(1UL, parameters.encodings.size());
  
      parameters.encodings[0].max_bitrate_bps = bitrate;
      return channel_->SetRtpSendParameters(ssrc, parameters).ok();
    }
  
    void SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) {
      cricket::AudioSendParameters send_parameters;
      send_parameters.codecs.push_back(codec);
      send_parameters.max_bandwidth_bps = bitrate;
      SetSendParameters(send_parameters);
    }
  
    void CheckSendCodecBitrate(int32_t ssrc,
                               const char expected_name[],
                               int expected_bitrate) {
      const auto& spec = GetSendStreamConfig(ssrc).send_codec_spec;
      EXPECT_EQ(expected_name, spec->format.name);
      EXPECT_EQ(expected_bitrate, spec->target_bitrate_bps);
    }
  
    absl::optional<int> GetCodecBitrate(int32_t ssrc) {
      return GetSendStreamConfig(ssrc).send_codec_spec->target_bitrate_bps;
    }
  
    const absl::optional<std::string>& GetAudioNetworkAdaptorConfig(
        int32_t ssrc) {
      return GetSendStreamConfig(ssrc).audio_network_adaptor_config;
    }
  
    void SetAndExpectMaxBitrate(const cricket::AudioCodec& codec,
                                int global_max,
                                int stream_max,
                                bool expected_result,
                                int expected_codec_bitrate) {
      // Clear the bitrate limit from the previous test case.
      EXPECT_TRUE(SetMaxBitrateForStream(kSsrcX, -1));
  
      // Attempt to set the requested bitrate limits.
      SetGlobalMaxBitrate(codec, global_max);
      EXPECT_EQ(expected_result, SetMaxBitrateForStream(kSsrcX, stream_max));
  
      // Verify that reading back the parameters gives results
      // consistent with the Set() result.
      webrtc::RtpParameters resulting_parameters =
          channel_->GetRtpSendParameters(kSsrcX);
      EXPECT_EQ(1UL, resulting_parameters.encodings.size());
      EXPECT_EQ(expected_result ? stream_max : -1,
                resulting_parameters.encodings[0].max_bitrate_bps);
  
      // Verify that the codec settings have the expected bitrate.
      EXPECT_EQ(expected_codec_bitrate, GetCodecBitrate(kSsrcX));
    }
  
    void SetSendCodecsShouldWorkForBitrates(const char* min_bitrate_kbps,
                                            int expected_min_bitrate_bps,
                                            const char* start_bitrate_kbps,
                                            int expected_start_bitrate_bps,
                                            const char* max_bitrate_kbps,
                                            int expected_max_bitrate_bps) {
      EXPECT_TRUE(SetupSendStream());
      auto& codecs = send_parameters_.codecs;
      codecs.clear();
      codecs.push_back(kOpusCodec);
      codecs[0].params[cricket::kCodecParamMinBitrate] = min_bitrate_kbps;
      codecs[0].params[cricket::kCodecParamStartBitrate] = start_bitrate_kbps;
      codecs[0].params[cricket::kCodecParamMaxBitrate] = max_bitrate_kbps;
      EXPECT_CALL(*call_.GetMockTransportControllerSend(),
                  SetSdpBitrateParameters(
                      AllOf(Field(&BitrateConstraints::min_bitrate_bps,
                                  expected_min_bitrate_bps),
                            Field(&BitrateConstraints::start_bitrate_bps,
                                  expected_start_bitrate_bps),
                            Field(&BitrateConstraints::max_bitrate_bps,
                                  expected_max_bitrate_bps))));
  
      SetSendParameters(send_parameters_);
    }
  
    void TestSetSendRtpHeaderExtensions(const std::string& ext) {
      EXPECT_TRUE(SetupSendStream());
  
      // Ensure extensions are off by default.
      EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size());
  
      // Ensure unknown extensions won't cause an error.
      send_parameters_.extensions.push_back(
          webrtc::RtpExtension("urn:ietf:params:unknownextention", 1));
      SetSendParameters(send_parameters_);
      EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size());
  
      // Ensure extensions stay off with an empty list of headers.
      send_parameters_.extensions.clear();
      SetSendParameters(send_parameters_);
      EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size());
  
      // Ensure extension is set properly.
      const int id = 1;
      send_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id));
      SetSendParameters(send_parameters_);
      EXPECT_EQ(1u, GetSendStreamConfig(kSsrcX).rtp.extensions.size());
      EXPECT_EQ(ext, GetSendStreamConfig(kSsrcX).rtp.extensions[0].uri);
      EXPECT_EQ(id, GetSendStreamConfig(kSsrcX).rtp.extensions[0].id);
  
      // Ensure extension is set properly on new stream.
      EXPECT_TRUE(
          channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcY)));
      EXPECT_NE(call_.GetAudioSendStream(kSsrcX),
                call_.GetAudioSendStream(kSsrcY));
      EXPECT_EQ(1u, GetSendStreamConfig(kSsrcY).rtp.extensions.size());
      EXPECT_EQ(ext, GetSendStreamConfig(kSsrcY).rtp.extensions[0].uri);
      EXPECT_EQ(id, GetSendStreamConfig(kSsrcY).rtp.extensions[0].id);
  
      // Ensure all extensions go back off with an empty list.
      send_parameters_.codecs.push_back(kPcmuCodec);
      send_parameters_.extensions.clear();
      SetSendParameters(send_parameters_);
      EXPECT_EQ(0u, GetSendStreamConfig(kSsrcX).rtp.extensions.size());
      EXPECT_EQ(0u, GetSendStreamConfig(kSsrcY).rtp.extensions.size());
    }
  
    void TestSetRecvRtpHeaderExtensions(const std::string& ext) {
      EXPECT_TRUE(SetupRecvStream());
  
      // Ensure extensions are off by default.
      EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size());
  
      // Ensure unknown extensions won't cause an error.
      recv_parameters_.extensions.push_back(
          webrtc::RtpExtension("urn:ietf:params:unknownextention", 1));
      EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
      EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size());
  
      // Ensure extensions stay off with an empty list of headers.
      recv_parameters_.extensions.clear();
      EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
      EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size());
  
      // Ensure extension is set properly.
      const int id = 2;
      recv_parameters_.extensions.push_back(webrtc::RtpExtension(ext, id));
      EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
      EXPECT_EQ(1u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size());
      EXPECT_EQ(ext, GetRecvStreamConfig(kSsrcX).rtp.extensions[0].uri);
      EXPECT_EQ(id, GetRecvStreamConfig(kSsrcX).rtp.extensions[0].id);
  
      // Ensure extension is set properly on new stream.
      EXPECT_TRUE(AddRecvStream(kSsrcY));
      EXPECT_NE(call_.GetAudioReceiveStream(kSsrcX),
                call_.GetAudioReceiveStream(kSsrcY));
      EXPECT_EQ(1u, GetRecvStreamConfig(kSsrcY).rtp.extensions.size());
      EXPECT_EQ(ext, GetRecvStreamConfig(kSsrcY).rtp.extensions[0].uri);
      EXPECT_EQ(id, GetRecvStreamConfig(kSsrcY).rtp.extensions[0].id);
  
      // Ensure all extensions go back off with an empty list.
      recv_parameters_.extensions.clear();
      EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
      EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcX).rtp.extensions.size());
      EXPECT_EQ(0u, GetRecvStreamConfig(kSsrcY).rtp.extensions.size());
    }
  
    webrtc::AudioSendStream::Stats GetAudioSendStreamStats() const {
      webrtc::AudioSendStream::Stats stats;
      stats.local_ssrc = 12;
      stats.bytes_sent = 345;
      stats.packets_sent = 678;
      stats.packets_lost = 9012;
      stats.fraction_lost = 34.56f;
      stats.codec_name = "codec_name_send";
      stats.codec_payload_type = 42;
      stats.ext_seqnum = 789;
      stats.jitter_ms = 12;
      stats.rtt_ms = 345;
      stats.audio_level = 678;
      stats.apm_statistics.delay_median_ms = 234;
      stats.apm_statistics.delay_standard_deviation_ms = 567;
      stats.apm_statistics.echo_return_loss = 890;
      stats.apm_statistics.echo_return_loss_enhancement = 1234;
      stats.apm_statistics.residual_echo_likelihood = 0.432f;
      stats.apm_statistics.residual_echo_likelihood_recent_max = 0.6f;
      stats.ana_statistics.bitrate_action_counter = 321;
      stats.ana_statistics.channel_action_counter = 432;
      stats.ana_statistics.dtx_action_counter = 543;
      stats.ana_statistics.fec_action_counter = 654;
      stats.ana_statistics.frame_length_increase_counter = 765;
      stats.ana_statistics.frame_length_decrease_counter = 876;
      stats.ana_statistics.uplink_packet_loss_fraction = 987.0;
      stats.typing_noise_detected = true;
      return stats;
    }
    void SetAudioSendStreamStats() {
      for (auto* s : call_.GetAudioSendStreams()) {
        s->SetStats(GetAudioSendStreamStats());
      }
    }
    void VerifyVoiceSenderInfo(const cricket::VoiceSenderInfo& info,
                               bool is_sending) {
      const auto stats = GetAudioSendStreamStats();
      EXPECT_EQ(info.ssrc(), stats.local_ssrc);
      EXPECT_EQ(info.bytes_sent, stats.bytes_sent);
      EXPECT_EQ(info.packets_sent, stats.packets_sent);
      EXPECT_EQ(info.packets_lost, stats.packets_lost);
      EXPECT_EQ(info.fraction_lost, stats.fraction_lost);
      EXPECT_EQ(info.codec_name, stats.codec_name);
      EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type);
      EXPECT_EQ(info.ext_seqnum, stats.ext_seqnum);
      EXPECT_EQ(info.jitter_ms, stats.jitter_ms);
      EXPECT_EQ(info.rtt_ms, stats.rtt_ms);
      EXPECT_EQ(info.audio_level, stats.audio_level);
      EXPECT_EQ(info.apm_statistics.delay_median_ms,
                stats.apm_statistics.delay_median_ms);
      EXPECT_EQ(info.apm_statistics.delay_standard_deviation_ms,
                stats.apm_statistics.delay_standard_deviation_ms);
      EXPECT_EQ(info.apm_statistics.echo_return_loss,
                stats.apm_statistics.echo_return_loss);
      EXPECT_EQ(info.apm_statistics.echo_return_loss_enhancement,
                stats.apm_statistics.echo_return_loss_enhancement);
      EXPECT_EQ(info.apm_statistics.residual_echo_likelihood,
                stats.apm_statistics.residual_echo_likelihood);
      EXPECT_EQ(info.apm_statistics.residual_echo_likelihood_recent_max,
                stats.apm_statistics.residual_echo_likelihood_recent_max);
      EXPECT_EQ(info.ana_statistics.bitrate_action_counter,
                stats.ana_statistics.bitrate_action_counter);
      EXPECT_EQ(info.ana_statistics.channel_action_counter,
                stats.ana_statistics.channel_action_counter);
      EXPECT_EQ(info.ana_statistics.dtx_action_counter,
                stats.ana_statistics.dtx_action_counter);
      EXPECT_EQ(info.ana_statistics.fec_action_counter,
                stats.ana_statistics.fec_action_counter);
      EXPECT_EQ(info.ana_statistics.frame_length_increase_counter,
                stats.ana_statistics.frame_length_increase_counter);
      EXPECT_EQ(info.ana_statistics.frame_length_decrease_counter,
                stats.ana_statistics.frame_length_decrease_counter);
      EXPECT_EQ(info.ana_statistics.uplink_packet_loss_fraction,
                stats.ana_statistics.uplink_packet_loss_fraction);
      EXPECT_EQ(info.typing_noise_detected,
                stats.typing_noise_detected && is_sending);
    }
  
    webrtc::AudioReceiveStream::Stats GetAudioReceiveStreamStats() const {
      webrtc::AudioReceiveStream::Stats stats;
      stats.remote_ssrc = 123;
      stats.bytes_rcvd = 456;
      stats.packets_rcvd = 768;
      stats.packets_lost = 101;
      stats.codec_name = "codec_name_recv";
      stats.codec_payload_type = 42;
      stats.ext_seqnum = 678;
      stats.jitter_ms = 901;
      stats.jitter_buffer_ms = 234;
      stats.jitter_buffer_preferred_ms = 567;
      stats.delay_estimate_ms = 890;
      stats.audio_level = 1234;
      stats.total_samples_received = 5678901;
      stats.concealed_samples = 234;
      stats.concealment_events = 12;
      stats.jitter_buffer_delay_seconds = 34;
      stats.jitter_buffer_emitted_count = 77;
      stats.expand_rate = 5.67f;
      stats.speech_expand_rate = 8.90f;
      stats.secondary_decoded_rate = 1.23f;
      stats.secondary_discarded_rate = 0.12f;
      stats.accelerate_rate = 4.56f;
      stats.preemptive_expand_rate = 7.89f;
      stats.decoding_calls_to_silence_generator = 12;
      stats.decoding_calls_to_neteq = 345;
      stats.decoding_normal = 67890;
      stats.decoding_plc = 1234;
      stats.decoding_cng = 5678;
      stats.decoding_plc_cng = 9012;
      stats.decoding_muted_output = 3456;
      stats.capture_start_ntp_time_ms = 7890;
      return stats;
    }
    void SetAudioReceiveStreamStats() {
      for (auto* s : call_.GetAudioReceiveStreams()) {
        s->SetStats(GetAudioReceiveStreamStats());
      }
    }
    void VerifyVoiceReceiverInfo(const cricket::VoiceReceiverInfo& info) {
      const auto stats = GetAudioReceiveStreamStats();
      EXPECT_EQ(info.ssrc(), stats.remote_ssrc);
      EXPECT_EQ(info.bytes_rcvd, stats.bytes_rcvd);
      EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_rcvd),
                stats.packets_rcvd);
      EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_lost),
                stats.packets_lost);
      EXPECT_EQ(info.codec_name, stats.codec_name);
      EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type);
      EXPECT_EQ(rtc::checked_cast<unsigned int>(info.ext_seqnum),
                stats.ext_seqnum);
      EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_ms), stats.jitter_ms);
      EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_ms),
                stats.jitter_buffer_ms);
      EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_preferred_ms),
                stats.jitter_buffer_preferred_ms);
      EXPECT_EQ(rtc::checked_cast<unsigned int>(info.delay_estimate_ms),
                stats.delay_estimate_ms);
      EXPECT_EQ(info.audio_level, stats.audio_level);
      EXPECT_EQ(info.total_samples_received, stats.total_samples_received);
      EXPECT_EQ(info.concealed_samples, stats.concealed_samples);
      EXPECT_EQ(info.concealment_events, stats.concealment_events);
      EXPECT_EQ(info.jitter_buffer_delay_seconds,
                stats.jitter_buffer_delay_seconds);
      EXPECT_EQ(info.jitter_buffer_emitted_count,
                stats.jitter_buffer_emitted_count);
      EXPECT_EQ(info.expand_rate, stats.expand_rate);
      EXPECT_EQ(info.speech_expand_rate, stats.speech_expand_rate);
      EXPECT_EQ(info.secondary_decoded_rate, stats.secondary_decoded_rate);
      EXPECT_EQ(info.secondary_discarded_rate, stats.secondary_discarded_rate);
      EXPECT_EQ(info.accelerate_rate, stats.accelerate_rate);
      EXPECT_EQ(info.preemptive_expand_rate, stats.preemptive_expand_rate);
      EXPECT_EQ(info.decoding_calls_to_silence_generator,
                stats.decoding_calls_to_silence_generator);
      EXPECT_EQ(info.decoding_calls_to_neteq, stats.decoding_calls_to_neteq);
      EXPECT_EQ(info.decoding_normal, stats.decoding_normal);
      EXPECT_EQ(info.decoding_plc, stats.decoding_plc);
      EXPECT_EQ(info.decoding_cng, stats.decoding_cng);
      EXPECT_EQ(info.decoding_plc_cng, stats.decoding_plc_cng);
      EXPECT_EQ(info.decoding_muted_output, stats.decoding_muted_output);
      EXPECT_EQ(info.capture_start_ntp_time_ms, stats.capture_start_ntp_time_ms);
    }
    void VerifyVoiceSendRecvCodecs(const cricket::VoiceMediaInfo& info) const {
      EXPECT_EQ(send_parameters_.codecs.size(), info.send_codecs.size());
      for (const cricket::AudioCodec& codec : send_parameters_.codecs) {
        ASSERT_EQ(info.send_codecs.count(codec.id), 1U);
        EXPECT_EQ(info.send_codecs.find(codec.id)->second,
                  codec.ToCodecParameters());
      }
      EXPECT_EQ(recv_parameters_.codecs.size(), info.receive_codecs.size());
      for (const cricket::AudioCodec& codec : recv_parameters_.codecs) {
        ASSERT_EQ(info.receive_codecs.count(codec.id), 1U);
        EXPECT_EQ(info.receive_codecs.find(codec.id)->second,
                  codec.ToCodecParameters());
      }
    }
  
    void VerifyGainControlEnabledCorrectly() {
      EXPECT_TRUE(apm_config_.gain_controller1.enabled);
      EXPECT_EQ(kDefaultAgcMode, apm_config_.gain_controller1.mode);
      EXPECT_EQ(0, apm_config_.gain_controller1.analog_level_minimum);
      EXPECT_EQ(255, apm_config_.gain_controller1.analog_level_maximum);
    }
  
    void VerifyGainControlDefaultSettings() {
      EXPECT_EQ(3, apm_config_.gain_controller1.target_level_dbfs);
      EXPECT_EQ(9, apm_config_.gain_controller1.compression_gain_db);
      EXPECT_TRUE(apm_config_.gain_controller1.enable_limiter);
    }
  
    bool IsEchoCancellationEnabled() {
      return apm_config_.echo_canceller.enabled;
    }
  
    bool IsHighPassFilterEnabled() {
      return apm_config_.high_pass_filter.enabled;
    }
  
    bool IsTypingDetectionEnabled() {
      return apm_config_.voice_detection.enabled;
    }
  
   protected:
    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
    StrictMock<webrtc::test::MockAudioDeviceModule> adm_;
    rtc::scoped_refptr<StrictMock<webrtc::test::MockAudioProcessing>> apm_;
    webrtc::test::MockNoiseSuppression& apm_ns_;
    cricket::FakeCall call_;
    std::unique_ptr<cricket::WebRtcVoiceEngine> engine_;
    cricket::VoiceMediaChannel* channel_ = nullptr;
    cricket::AudioSendParameters send_parameters_;
    cricket::AudioRecvParameters recv_parameters_;
    FakeAudioSource fake_source_;
    webrtc::AudioProcessing::Config apm_config_;
  
   private:
    webrtc::test::ScopedFieldTrials override_field_trials_;
  };
  
  // Tests that we can create and destroy a channel.
  TEST_F(WebRtcVoiceEngineTestFake, CreateMediaChannel) {
    EXPECT_TRUE(SetupChannel());
  }
  
  // Test that we can add a send stream and that it has the correct defaults.
  TEST_F(WebRtcVoiceEngineTestFake, CreateSendStream) {
    EXPECT_TRUE(SetupChannel());
    EXPECT_TRUE(
        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
    const webrtc::AudioSendStream::Config& config = GetSendStreamConfig(kSsrcX);
    EXPECT_EQ(kSsrcX, config.rtp.ssrc);
    EXPECT_EQ("", config.rtp.c_name);
    EXPECT_EQ(0u, config.rtp.extensions.size());
    EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
              config.send_transport);
  }
  
  // Test that we can add a receive stream and that it has the correct defaults.
  TEST_F(WebRtcVoiceEngineTestFake, CreateRecvStream) {
    EXPECT_TRUE(SetupChannel());
    EXPECT_TRUE(AddRecvStream(kSsrcX));
    const webrtc::AudioReceiveStream::Config& config =
        GetRecvStreamConfig(kSsrcX);
    EXPECT_EQ(kSsrcX, config.rtp.remote_ssrc);
    EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc);
    EXPECT_FALSE(config.rtp.transport_cc);
    EXPECT_EQ(0u, config.rtp.extensions.size());
    EXPECT_EQ(static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_),
              config.rtcp_send_transport);
    EXPECT_EQ("", config.sync_group);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, OpusSupportsTransportCc) {
    const std::vector<cricket::AudioCodec>& codecs = engine_->send_codecs();
    bool opus_found = false;
    for (const cricket::AudioCodec& codec : codecs) {
      if (codec.name == "opus") {
        EXPECT_TRUE(HasTransportCc(codec));
        opus_found = true;
      }
    }
    EXPECT_TRUE(opus_found);
  }
  
  // Test that we set our inbound codecs properly, including changing PT.
  TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecs) {
    EXPECT_TRUE(SetupChannel());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs.push_back(kTelephoneEventCodec1);
    parameters.codecs.push_back(kTelephoneEventCodec2);
    parameters.codecs[0].id = 106;  // collide with existing CN 32k
    parameters.codecs[2].id = 126;
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
    EXPECT_TRUE(AddRecvStream(kSsrcX));
    EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
                (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
                    {{0, {"PCMU", 8000, 1}},
                     {106, {"ISAC", 16000, 1}},
                     {126, {"telephone-event", 8000, 1}},
                     {107, {"telephone-event", 32000, 1}}})));
  }
  
  // Test that we fail to set an unknown inbound codec.
  TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsUnsupportedCodec) {
    EXPECT_TRUE(SetupChannel());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(cricket::AudioCodec(127, "XYZ", 32000, 0, 1));
    EXPECT_FALSE(channel_->SetRecvParameters(parameters));
  }
  
  // Test that we fail if we have duplicate types in the inbound list.
  TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsDuplicatePayloadType) {
    EXPECT_TRUE(SetupChannel());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kCn16000Codec);
    parameters.codecs[1].id = kIsacCodec.id;
    EXPECT_FALSE(channel_->SetRecvParameters(parameters));
  }
  
  // Test that we can decode OPUS without stereo parameters.
  TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpusNoStereo) {
    EXPECT_TRUE(SetupChannel());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs.push_back(kOpusCodec);
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
    EXPECT_TRUE(AddRecvStream(kSsrcX));
    EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
                (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
                    {{0, {"PCMU", 8000, 1}},
                     {103, {"ISAC", 16000, 1}},
                     {111, {"opus", 48000, 2}}})));
  }
  
  // Test that we can decode OPUS with stereo = 0.
  TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus0Stereo) {
    EXPECT_TRUE(SetupChannel());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[2].params["stereo"] = "0";
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
    EXPECT_TRUE(AddRecvStream(kSsrcX));
    EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
                (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
                    {{0, {"PCMU", 8000, 1}},
                     {103, {"ISAC", 16000, 1}},
                     {111, {"opus", 48000, 2, {{"stereo", "0"}}}}})));
  }
  
  // Test that we can decode OPUS with stereo = 1.
  TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithOpus1Stereo) {
    EXPECT_TRUE(SetupChannel());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[2].params["stereo"] = "1";
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
    EXPECT_TRUE(AddRecvStream(kSsrcX));
    EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
                (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
                    {{0, {"PCMU", 8000, 1}},
                     {103, {"ISAC", 16000, 1}},
                     {111, {"opus", 48000, 2, {{"stereo", "1"}}}}})));
  }
  
  // Test that changes to recv codecs are applied to all streams.
  TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWithMultipleStreams) {
    EXPECT_TRUE(SetupChannel());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs.push_back(kTelephoneEventCodec1);
    parameters.codecs.push_back(kTelephoneEventCodec2);
    parameters.codecs[0].id = 106;  // collide with existing CN 32k
    parameters.codecs[2].id = 126;
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
    for (const auto& ssrc : {kSsrcX, kSsrcY}) {
      EXPECT_TRUE(AddRecvStream(ssrc));
      EXPECT_THAT(GetRecvStreamConfig(ssrc).decoder_map,
                  (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
                      {{0, {"PCMU", 8000, 1}},
                       {106, {"ISAC", 16000, 1}},
                       {126, {"telephone-event", 8000, 1}},
                       {107, {"telephone-event", 32000, 1}}})));
    }
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
    EXPECT_TRUE(SetupRecvStream());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs[0].id = 106;  // collide with existing CN 32k
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
  
    const auto& dm = GetRecvStreamConfig(kSsrcX).decoder_map;
    ASSERT_EQ(1u, dm.count(106));
    EXPECT_EQ(webrtc::SdpAudioFormat("isac", 16000, 1), dm.at(106));
  }
  
  // Test that we can apply the same set of codecs again while playing.
  TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsWhilePlaying) {
    EXPECT_TRUE(SetupRecvStream());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kCn16000Codec);
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
    channel_->SetPlayout(true);
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
  
    // Remapping a payload type to a different codec should fail.
    parameters.codecs[0] = kOpusCodec;
    parameters.codecs[0].id = kIsacCodec.id;
    EXPECT_FALSE(channel_->SetRecvParameters(parameters));
    EXPECT_TRUE(GetRecvStream(kSsrcX).started());
  }
  
  // Test that we can add a codec while playing.
  TEST_F(WebRtcVoiceEngineTestFake, AddRecvCodecsWhilePlaying) {
    EXPECT_TRUE(SetupRecvStream());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kCn16000Codec);
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
    channel_->SetPlayout(true);
  
    parameters.codecs.push_back(kOpusCodec);
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
    EXPECT_TRUE(GetRecvStream(kSsrcX).started());
  }
  
  // Test that we accept adding the same codec with a different payload type.
  // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5847
  TEST_F(WebRtcVoiceEngineTestFake, ChangeRecvCodecPayloadType) {
    EXPECT_TRUE(SetupRecvStream());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
  
    ++parameters.codecs[0].id;
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetSendBandwidthAuto) {
    EXPECT_TRUE(SetupSendStream());
  
    // Test that when autobw is enabled, bitrate is kept as the default
    // value. autobw is enabled for the following tests because the target
    // bitrate is <= 0.
  
    // ISAC, default bitrate == 32000.
    TestMaxSendBandwidth(kIsacCodec, 0, true, 32000);
  
    // PCMU, default bitrate == 64000.
    TestMaxSendBandwidth(kPcmuCodec, -1, true, 64000);
  
    // opus, default bitrate == 32000 in mono.
    TestMaxSendBandwidth(kOpusCodec, -1, true, 32000);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCaller) {
    EXPECT_TRUE(SetupSendStream());
  
    // ISAC, default bitrate == 32000.
    TestMaxSendBandwidth(kIsacCodec, 16000, true, 16000);
    // Rates above the max (56000) should be capped.
    TestMaxSendBandwidth(kIsacCodec, 100000, true, 32000);
  
    // opus, default bitrate == 64000.
    TestMaxSendBandwidth(kOpusCodec, 96000, true, 96000);
    TestMaxSendBandwidth(kOpusCodec, 48000, true, 48000);
    // Rates above the max (510000) should be capped.
    TestMaxSendBandwidth(kOpusCodec, 600000, true, 510000);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthFixedRateAsCaller) {
    EXPECT_TRUE(SetupSendStream());
  
    // Test that we can only set a maximum bitrate for a fixed-rate codec
    // if it's bigger than the fixed rate.
  
    // PCMU, fixed bitrate == 64000.
    TestMaxSendBandwidth(kPcmuCodec, 0, true, 64000);
    TestMaxSendBandwidth(kPcmuCodec, 1, false, 64000);
    TestMaxSendBandwidth(kPcmuCodec, 128000, true, 64000);
    TestMaxSendBandwidth(kPcmuCodec, 32000, false, 64000);
    TestMaxSendBandwidth(kPcmuCodec, 64000, true, 64000);
    TestMaxSendBandwidth(kPcmuCodec, 63999, false, 64000);
    TestMaxSendBandwidth(kPcmuCodec, 64001, true, 64000);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) {
    EXPECT_TRUE(SetupChannel());
    const int kDesiredBitrate = 128000;
    cricket::AudioSendParameters parameters;
    parameters.codecs = engine_->send_codecs();
    parameters.max_bandwidth_bps = kDesiredBitrate;
    SetSendParameters(parameters);
  
    EXPECT_TRUE(
        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
  
    EXPECT_EQ(kDesiredBitrate, GetCodecBitrate(kSsrcX));
  }
  
  // Test that bitrate cannot be set for CBR codecs.
  // Bitrate is ignored if it is higher than the fixed bitrate.
  // Bitrate less then the fixed bitrate is an error.
  TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthCbr) {
    EXPECT_TRUE(SetupSendStream());
  
    // PCMU, default bitrate == 64000.
    SetSendParameters(send_parameters_);
    EXPECT_EQ(64000, GetCodecBitrate(kSsrcX));
  
    send_parameters_.max_bandwidth_bps = 128000;
    SetSendParameters(send_parameters_);
    EXPECT_EQ(64000, GetCodecBitrate(kSsrcX));
  
    send_parameters_.max_bandwidth_bps = 128;
    EXPECT_FALSE(channel_->SetSendParameters(send_parameters_));
    EXPECT_EQ(64000, GetCodecBitrate(kSsrcX));
  }
  
  // Test that the per-stream bitrate limit and the global
  // bitrate limit both apply.
  TEST_F(WebRtcVoiceEngineTestFake, SetMaxBitratePerStream) {
    EXPECT_TRUE(SetupSendStream());
  
    // opus, default bitrate == 32000.
    SetAndExpectMaxBitrate(kOpusCodec, 0, 0, true, 32000);
    SetAndExpectMaxBitrate(kOpusCodec, 48000, 0, true, 48000);
    SetAndExpectMaxBitrate(kOpusCodec, 48000, 64000, true, 48000);
    SetAndExpectMaxBitrate(kOpusCodec, 64000, 48000, true, 48000);
  
    // CBR codecs allow both maximums to exceed the bitrate.
    SetAndExpectMaxBitrate(kPcmuCodec, 0, 0, true, 64000);
    SetAndExpectMaxBitrate(kPcmuCodec, 64001, 0, true, 64000);
    SetAndExpectMaxBitrate(kPcmuCodec, 0, 64001, true, 64000);
    SetAndExpectMaxBitrate(kPcmuCodec, 64001, 64001, true, 64000);
  
    // CBR codecs don't allow per stream maximums to be too low.
    SetAndExpectMaxBitrate(kPcmuCodec, 0, 63999, false, 64000);
    SetAndExpectMaxBitrate(kPcmuCodec, 64001, 63999, false, 64000);
  }
  
  // Test that an attempt to set RtpParameters for a stream that does not exist
  // fails.
  TEST_F(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) {
    EXPECT_TRUE(SetupChannel());
    webrtc::RtpParameters nonexistent_parameters =
        channel_->GetRtpSendParameters(kSsrcX);
    EXPECT_EQ(0u, nonexistent_parameters.encodings.size());
  
    nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters());
    EXPECT_FALSE(
        channel_->SetRtpSendParameters(kSsrcX, nonexistent_parameters).ok());
  }
  
  TEST_F(WebRtcVoiceEngineTestFake,
         CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) {
    // This test verifies that setting RtpParameters succeeds only if
    // the structure contains exactly one encoding.
    // TODO(skvlad): Update this test when we start supporting setting parameters
    // for each encoding individually.
  
    EXPECT_TRUE(SetupSendStream());
    webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
    // Two or more encodings should result in failure.
    parameters.encodings.push_back(webrtc::RtpEncodingParameters());
    EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
    // Zero encodings should also fail.
    parameters.encodings.clear();
    EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
  }
  
  // Changing the SSRC through RtpParameters is not allowed.
  TEST_F(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) {
    EXPECT_TRUE(SetupSendStream());
    webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
    parameters.encodings[0].ssrc = 0xdeadbeef;
    EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
  }
  
  // Test that a stream will not be sending if its encoding is made
  // inactive through SetRtpSendParameters.
  TEST_F(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) {
    EXPECT_TRUE(SetupSendStream());
    SetSend(true);
    EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
    // Get current parameters and change "active" to false.
    webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
    ASSERT_EQ(1u, parameters.encodings.size());
    ASSERT_TRUE(parameters.encodings[0].active);
    parameters.encodings[0].active = false;
    EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
    EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
  
    // Now change it back to active and verify we resume sending.
    // This should occur even when other parameters are updated.
    parameters.encodings[0].active = true;
    parameters.encodings[0].max_bitrate_bps = absl::optional<int>(6000);
    EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
    EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
  }
  
  // Test that SetRtpSendParameters configures the correct encoding channel for
  // each SSRC.
  TEST_F(WebRtcVoiceEngineTestFake, RtpParametersArePerStream) {
    SetupForMultiSendStream();
    // Create send streams.
    for (uint32_t ssrc : kSsrcs4) {
      EXPECT_TRUE(
          channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
    }
    // Configure one stream to be limited by the stream config, another to be
    // limited by the global max, and the third one with no per-stream limit
    // (still subject to the global limit).
    SetGlobalMaxBitrate(kOpusCodec, 32000);
    EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[0], 24000));
    EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[1], 48000));
    EXPECT_TRUE(SetMaxBitrateForStream(kSsrcs4[2], -1));
  
    EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0]));
    EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[1]));
    EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2]));
  
    // Remove the global cap; the streams should switch to their respective
    // maximums (or remain unchanged if there was no other limit on them.)
    SetGlobalMaxBitrate(kOpusCodec, -1);
    EXPECT_EQ(24000, GetCodecBitrate(kSsrcs4[0]));
    EXPECT_EQ(48000, GetCodecBitrate(kSsrcs4[1]));
    EXPECT_EQ(32000, GetCodecBitrate(kSsrcs4[2]));
  }
  
  // RTCRtpEncodingParameters.network_priority must be one of a few values
  // derived from the default priority, corresponding to very-low, low, medium,
  // or high.
  TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParametersInvalidNetworkPriority) {
    EXPECT_TRUE(SetupSendStream());
    webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
    EXPECT_EQ(1UL, parameters.encodings.size());
    EXPECT_EQ(webrtc::kDefaultBitratePriority,
              parameters.encodings[0].network_priority);
  
    double good_values[] = {0.5, 1.0, 2.0, 4.0};
    double bad_values[] = {-1.0, 0.0, 0.49, 0.51, 1.1, 3.99, 4.1, 5.0};
    for (auto it : good_values) {
      parameters.encodings[0].network_priority = it;
      EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
    }
    for (auto it : bad_values) {
      parameters.encodings[0].network_priority = it;
      EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
    }
  }
  
  // Test that GetRtpSendParameters returns the currently configured codecs.
  TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    SetSendParameters(parameters);
  
    webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
    ASSERT_EQ(2u, rtp_parameters.codecs.size());
    EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]);
    EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
  }
  
  // Test that GetRtpSendParameters returns the currently configured RTCP CNAME.
  TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersRtcpCname) {
    cricket::StreamParams params = cricket::StreamParams::CreateLegacy(kSsrcX);
    params.cname = "rtcpcname";
    EXPECT_TRUE(SetupSendStream(params));
  
    webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
    EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
  }
  
  TEST_F(WebRtcVoiceEngineTestFake,
         DetectRtpSendParameterHeaderExtensionsChange) {
    EXPECT_TRUE(SetupSendStream());
  
    webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
    rtp_parameters.header_extensions.emplace_back();
  
    EXPECT_NE(0u, rtp_parameters.header_extensions.size());
  
    webrtc::RTCError result =
        channel_->SetRtpSendParameters(kSsrcX, rtp_parameters);
    EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type());
  }
  
  // Test that GetRtpSendParameters returns an SSRC.
  TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) {
    EXPECT_TRUE(SetupSendStream());
    webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
    ASSERT_EQ(1u, rtp_parameters.encodings.size());
    EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc);
  }
  
  // Test that if we set/get parameters multiple times, we get the same results.
  TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    SetSendParameters(parameters);
  
    webrtc::RtpParameters initial_params = channel_->GetRtpSendParameters(kSsrcX);
  
    // We should be able to set the params we just got.
    EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, initial_params).ok());
  
    // ... And this shouldn't change the params returned by GetRtpSendParameters.
    webrtc::RtpParameters new_params = channel_->GetRtpSendParameters(kSsrcX);
    EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(kSsrcX));
  }
  
  // Test that max_bitrate_bps in send stream config gets updated correctly when
  // SetRtpSendParameters is called.
  TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) {
    webrtc::test::ScopedFieldTrials override_field_trials(
        "WebRTC-Audio-SendSideBwe/Enabled/");
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters send_parameters;
    send_parameters.codecs.push_back(kOpusCodec);
    SetSendParameters(send_parameters);
  
    webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
    // Expect empty on parameters.encodings[0].max_bitrate_bps;
    EXPECT_FALSE(rtp_parameters.encodings[0].max_bitrate_bps);
  
    constexpr int kMaxBitrateBps = 6000;
    rtp_parameters.encodings[0].max_bitrate_bps = kMaxBitrateBps;
    EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
  
    const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps;
    EXPECT_EQ(max_bitrate, kMaxBitrateBps);
  }
  
  // Tests that when RTCRtpEncodingParameters.bitrate_priority gets set to
  // a value <= 0, setting the parameters returns false.
  TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParameterInvalidBitratePriority) {
    EXPECT_TRUE(SetupSendStream());
    webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
    EXPECT_EQ(1UL, rtp_parameters.encodings.size());
    EXPECT_EQ(webrtc::kDefaultBitratePriority,
              rtp_parameters.encodings[0].bitrate_priority);
  
    rtp_parameters.encodings[0].bitrate_priority = 0;
    EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
    rtp_parameters.encodings[0].bitrate_priority = -1.0;
    EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
  }
  
  // Test that the bitrate_priority in the send stream config gets updated when
  // SetRtpSendParameters is set for the VoiceMediaChannel.
  TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesBitratePriority) {
    EXPECT_TRUE(SetupSendStream());
    webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
  
    EXPECT_EQ(1UL, rtp_parameters.encodings.size());
    EXPECT_EQ(webrtc::kDefaultBitratePriority,
              rtp_parameters.encodings[0].bitrate_priority);
    double new_bitrate_priority = 2.0;
    rtp_parameters.encodings[0].bitrate_priority = new_bitrate_priority;
    EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
  
    // The priority should get set for both the audio channel's rtp parameters
    // and the audio send stream's audio config.
    EXPECT_EQ(
        new_bitrate_priority,
        channel_->GetRtpSendParameters(kSsrcX).encodings[0].bitrate_priority);
    EXPECT_EQ(new_bitrate_priority, GetSendStreamConfig(kSsrcX).bitrate_priority);
  }
  
  // Test that GetRtpReceiveParameters returns the currently configured codecs.
  TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersCodecs) {
    EXPECT_TRUE(SetupRecvStream());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
  
    webrtc::RtpParameters rtp_parameters =
        channel_->GetRtpReceiveParameters(kSsrcX);
    ASSERT_EQ(2u, rtp_parameters.codecs.size());
    EXPECT_EQ(kIsacCodec.ToCodecParameters(), rtp_parameters.codecs[0]);
    EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
  }
  
  // Test that GetRtpReceiveParameters returns an SSRC.
  TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersSsrc) {
    EXPECT_TRUE(SetupRecvStream());
    webrtc::RtpParameters rtp_parameters =
        channel_->GetRtpReceiveParameters(kSsrcX);
    ASSERT_EQ(1u, rtp_parameters.encodings.size());
    EXPECT_EQ(kSsrcX, rtp_parameters.encodings[0].ssrc);
  }
  
  // Test that if we set/get parameters multiple times, we get the same results.
  TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpReceiveParameters) {
    EXPECT_TRUE(SetupRecvStream());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
  
    webrtc::RtpParameters initial_params =
        channel_->GetRtpReceiveParameters(kSsrcX);
  
    // We should be able to set the params we just got.
    EXPECT_TRUE(channel_->SetRtpReceiveParameters(kSsrcX, initial_params));
  
    // ... And this shouldn't change the params returned by
    // GetRtpReceiveParameters.
    webrtc::RtpParameters new_params = channel_->GetRtpReceiveParameters(kSsrcX);
    EXPECT_EQ(initial_params, channel_->GetRtpReceiveParameters(kSsrcX));
  }
  
  // Test that GetRtpReceiveParameters returns parameters correctly when SSRCs
  // aren't signaled. It should return an empty "RtpEncodingParameters" when
  // configured to receive an unsignaled stream and no packets have been received
  // yet, and start returning the SSRC once a packet has been received.
  TEST_F(WebRtcVoiceEngineTestFake, GetRtpReceiveParametersWithUnsignaledSsrc) {
    ASSERT_TRUE(SetupChannel());
    // Call necessary methods to configure receiving a default stream as
    // soon as it arrives.
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
  
    // Call GetRtpReceiveParameters before configured to receive an unsignaled
    // stream. Should return nothing.
    EXPECT_EQ(webrtc::RtpParameters(), channel_->GetRtpReceiveParameters(0));
  
    // Set a sink for an unsignaled stream.
    std::unique_ptr<FakeAudioSink> fake_sink(new FakeAudioSink());
    // Value of "0" means "unsignaled stream".
    channel_->SetRawAudioSink(0, std::move(fake_sink));
  
    // Call GetRtpReceiveParameters before the SSRC is known. Value of "0"
    // in this method means "unsignaled stream".
    webrtc::RtpParameters rtp_parameters = channel_->GetRtpReceiveParameters(0);
    ASSERT_EQ(1u, rtp_parameters.encodings.size());
    EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
  
    // Receive PCMU packet (SSRC=1).
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
  
    // The |ssrc| member should still be unset.
    rtp_parameters = channel_->GetRtpReceiveParameters(0);
    ASSERT_EQ(1u, rtp_parameters.encodings.size());
    EXPECT_FALSE(rtp_parameters.encodings[0].ssrc);
  }
  
  // Test that we apply codecs properly.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs.push_back(kCn8000Codec);
    parameters.codecs[0].id = 96;
    parameters.codecs[0].bitrate = 22000;
    SetSendParameters(parameters);
    const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
    EXPECT_EQ(96, send_codec_spec.payload_type);
    EXPECT_EQ(22000, send_codec_spec.target_bitrate_bps);
    EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
    EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000);
    EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
    EXPECT_FALSE(channel_->CanInsertDtmf());
  }
  
  // Test that WebRtcVoiceEngine reconfigures, rather than recreates its
  // AudioSendStream.
  TEST_F(WebRtcVoiceEngineTestFake, DontRecreateSendStream) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs.push_back(kCn8000Codec);
    parameters.codecs[0].id = 96;
    parameters.codecs[0].bitrate = 48000;
    const int initial_num = call_.GetNumCreatedSendStreams();
    SetSendParameters(parameters);
    EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
    // Calling SetSendCodec again with same codec which is already set.
    // In this case media channel shouldn't send codec to VoE.
    SetSendParameters(parameters);
    EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
  }
  
  // TODO(ossu): Revisit if these tests need to be here, now that these kinds of
  // tests should be available in AudioEncoderOpusTest.
  
  // Test that if clockrate is not 48000 for opus, we fail.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 0;
    parameters.codecs[0].clockrate = 50000;
    EXPECT_FALSE(channel_->SetSendParameters(parameters));
  }
  
  // Test that if channels=0 for opus, we fail.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0ChannelsNoStereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 0;
    parameters.codecs[0].channels = 0;
    EXPECT_FALSE(channel_->SetSendParameters(parameters));
  }
  
  // Test that if channels=0 for opus, we fail.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 0;
    parameters.codecs[0].channels = 0;
    parameters.codecs[0].params["stereo"] = "1";
    EXPECT_FALSE(channel_->SetSendParameters(parameters));
  }
  
  // Test that if channel is 1 for opus and there's no stereo, we fail.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpus1ChannelNoStereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 0;
    parameters.codecs[0].channels = 1;
    EXPECT_FALSE(channel_->SetSendParameters(parameters));
  }
  
  // Test that if channel is 1 for opus and stereo=0, we fail.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 0;
    parameters.codecs[0].channels = 1;
    parameters.codecs[0].params["stereo"] = "0";
    EXPECT_FALSE(channel_->SetSendParameters(parameters));
  }
  
  // Test that if channel is 1 for opus and stereo=1, we fail.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 0;
    parameters.codecs[0].channels = 1;
    parameters.codecs[0].params["stereo"] = "1";
    EXPECT_FALSE(channel_->SetSendParameters(parameters));
  }
  
  // Test that with bitrate=0 and no stereo, bitrate is 32000.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0BitrateNoStereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 0;
    SetSendParameters(parameters);
    CheckSendCodecBitrate(kSsrcX, "opus", 32000);
  }
  
  // Test that with bitrate=0 and stereo=0, bitrate is 32000.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate0Stereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 0;
    parameters.codecs[0].params["stereo"] = "0";
    SetSendParameters(parameters);
    CheckSendCodecBitrate(kSsrcX, "opus", 32000);
  }
  
  // Test that with bitrate=invalid and stereo=0, bitrate is 32000.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate0Stereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].params["stereo"] = "0";
    // bitrate that's out of the range between 6000 and 510000 will be clamped.
    parameters.codecs[0].bitrate = 5999;
    SetSendParameters(parameters);
    CheckSendCodecBitrate(kSsrcX, "opus", 6000);
  
    parameters.codecs[0].bitrate = 510001;
    SetSendParameters(parameters);
    CheckSendCodecBitrate(kSsrcX, "opus", 510000);
  }
  
  // Test that with bitrate=0 and stereo=1, bitrate is 64000.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGood0Bitrate1Stereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 0;
    parameters.codecs[0].params["stereo"] = "1";
    SetSendParameters(parameters);
    CheckSendCodecBitrate(kSsrcX, "opus", 64000);
  }
  
  // Test that with bitrate=invalid and stereo=1, bitrate is 64000.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodXBitrate1Stereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].params["stereo"] = "1";
    // bitrate that's out of the range between 6000 and 510000 will be clamped.
    parameters.codecs[0].bitrate = 5999;
    SetSendParameters(parameters);
    CheckSendCodecBitrate(kSsrcX, "opus", 6000);
  
    parameters.codecs[0].bitrate = 510001;
    SetSendParameters(parameters);
    CheckSendCodecBitrate(kSsrcX, "opus", 510000);
  }
  
  // Test that with bitrate=N and stereo unset, bitrate is N.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 96000;
    SetSendParameters(parameters);
    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
    EXPECT_EQ(111, spec.payload_type);
    EXPECT_EQ(96000, spec.target_bitrate_bps);
    EXPECT_EQ("opus", spec.format.name);
    EXPECT_EQ(2u, spec.format.num_channels);
    EXPECT_EQ(48000, spec.format.clockrate_hz);
  }
  
  // Test that with bitrate=N and stereo=0, bitrate is N.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate0Stereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 30000;
    parameters.codecs[0].params["stereo"] = "0";
    SetSendParameters(parameters);
    CheckSendCodecBitrate(kSsrcX, "opus", 30000);
  }
  
  // Test that with bitrate=N and without any parameters, bitrate is N.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoParameters) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 30000;
    SetSendParameters(parameters);
    CheckSendCodecBitrate(kSsrcX, "opus", 30000);
  }
  
  // Test that with bitrate=N and stereo=1, bitrate is N.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrate1Stereo) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].bitrate = 30000;
    parameters.codecs[0].params["stereo"] = "1";
    SetSendParameters(parameters);
    CheckSendCodecBitrate(kSsrcX, "opus", 30000);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithBitrates) {
    SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
                                       200000);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithHighMaxBitrate) {
    SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "10000", 10000000);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake,
         SetSendCodecsWithoutBitratesUsesCorrectDefaults) {
    SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "", -1);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCapsMinAndStartBitrate) {
    SetSendCodecsShouldWorkForBitrates("-1", 0, "-100", -1, "", -1);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthForAudioDoesntAffectBwe) {
    SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
                                       200000);
    send_parameters_.max_bandwidth_bps = 100000;
    // Setting max bitrate should keep previous min bitrate
    // Setting max bitrate should not reset start bitrate.
    EXPECT_CALL(*call_.GetMockTransportControllerSend(),
                SetSdpBitrateParameters(
                    AllOf(Field(&BitrateConstraints::min_bitrate_bps, 100000),
                          Field(&BitrateConstraints::start_bitrate_bps, -1),
                          Field(&BitrateConstraints::max_bitrate_bps, 200000))));
    SetSendParameters(send_parameters_);
  }
  
  // Test that we can enable NACK with opus as callee.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
    EXPECT_TRUE(SetupRecvStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
        cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
    EXPECT_EQ(0, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
    SetSendParameters(parameters);
    // NACK should be enabled even with no send stream.
    EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
  
    EXPECT_TRUE(
        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
  }
  
  // Test that we can enable NACK on receive streams.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) {
    EXPECT_TRUE(SetupSendStream());
    EXPECT_TRUE(AddRecvStream(kSsrcY));
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
        cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
    EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
    SetSendParameters(parameters);
    EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
  }
  
  // Test that we can disable NACK on receive streams.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) {
    EXPECT_TRUE(SetupSendStream());
    EXPECT_TRUE(AddRecvStream(kSsrcY));
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kOpusCodec);
    parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
        cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
    SetSendParameters(parameters);
    EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
  
    parameters.codecs.clear();
    parameters.codecs.push_back(kOpusCodec);
    SetSendParameters(parameters);
    EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
  }
  
  // Test that NACK is enabled on a new receive stream.
  TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kCn16000Codec);
    parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
        cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
    SetSendParameters(parameters);
  
    EXPECT_TRUE(AddRecvStream(kSsrcY));
    EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
    EXPECT_TRUE(AddRecvStream(kSsrcZ));
    EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcZ).rtp.nack.rtp_history_ms);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, TransportCcCanBeEnabledAndDisabled) {
    EXPECT_TRUE(SetupChannel());
    cricket::AudioSendParameters send_parameters;
    send_parameters.codecs.push_back(kOpusCodec);
    EXPECT_TRUE(send_parameters.codecs[0].feedback_params.params().empty());
    SetSendParameters(send_parameters);
  
    cricket::AudioRecvParameters recv_parameters;
    recv_parameters.codecs.push_back(kIsacCodec);
    EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters));
    EXPECT_TRUE(AddRecvStream(kSsrcX));
    ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr);
    EXPECT_FALSE(
        call_.GetAudioReceiveStream(kSsrcX)->GetConfig().rtp.transport_cc);
  
    send_parameters.codecs = engine_->send_codecs();
    SetSendParameters(send_parameters);
    ASSERT_TRUE(call_.GetAudioReceiveStream(kSsrcX) != nullptr);
    EXPECT_TRUE(
        call_.GetAudioReceiveStream(kSsrcX)->GetConfig().rtp.transport_cc);
  }
  
  // Test that we can switch back and forth between Opus and ISAC with CN.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsIsacOpusSwitching) {
    EXPECT_TRUE(SetupSendStream());
  
    cricket::AudioSendParameters opus_parameters;
    opus_parameters.codecs.push_back(kOpusCodec);
    SetSendParameters(opus_parameters);
    {
      const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_EQ(111, spec.payload_type);
      EXPECT_STRCASEEQ("opus", spec.format.name.c_str());
    }
  
    cricket::AudioSendParameters isac_parameters;
    isac_parameters.codecs.push_back(kIsacCodec);
    isac_parameters.codecs.push_back(kCn16000Codec);
    isac_parameters.codecs.push_back(kOpusCodec);
    SetSendParameters(isac_parameters);
    {
      const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_EQ(103, spec.payload_type);
      EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
    }
  
    SetSendParameters(opus_parameters);
    {
      const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_EQ(111, spec.payload_type);
      EXPECT_STRCASEEQ("opus", spec.format.name.c_str());
    }
  }
  
  // Test that we handle various ways of specifying bitrate.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kIsacCodec);  // bitrate == 32000
    SetSendParameters(parameters);
    {
      const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_EQ(103, spec.payload_type);
      EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
      EXPECT_EQ(32000, spec.target_bitrate_bps);
    }
  
    parameters.codecs[0].bitrate = 0;  // bitrate == default
    SetSendParameters(parameters);
    {
      const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_EQ(103, spec.payload_type);
      EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
      EXPECT_EQ(32000, spec.target_bitrate_bps);
    }
    parameters.codecs[0].bitrate = 28000;  // bitrate == 28000
    SetSendParameters(parameters);
    {
      const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_EQ(103, spec.payload_type);
      EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
      EXPECT_EQ(28000, spec.target_bitrate_bps);
    }
  
    parameters.codecs[0] = kPcmuCodec;  // bitrate == 64000
    SetSendParameters(parameters);
    {
      const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_EQ(0, spec.payload_type);
      EXPECT_STRCASEEQ("PCMU", spec.format.name.c_str());
      EXPECT_EQ(64000, spec.target_bitrate_bps);
    }
  
    parameters.codecs[0].bitrate = 0;  // bitrate == default
    SetSendParameters(parameters);
    {
      const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_EQ(0, spec.payload_type);
      EXPECT_STREQ("PCMU", spec.format.name.c_str());
      EXPECT_EQ(64000, spec.target_bitrate_bps);
    }
  
    parameters.codecs[0] = kOpusCodec;
    parameters.codecs[0].bitrate = 0;  // bitrate == default
    SetSendParameters(parameters);
    {
      const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_EQ(111, spec.payload_type);
      EXPECT_STREQ("opus", spec.format.name.c_str());
      EXPECT_EQ(32000, spec.target_bitrate_bps);
    }
  }
  
  // Test that we fail if no codecs are specified.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    EXPECT_FALSE(channel_->SetSendParameters(parameters));
  }
  
  // Test that we can set send codecs even with telephone-event codec as the first
  // one on the list.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kTelephoneEventCodec1);
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs[0].id = 98;  // DTMF
    parameters.codecs[1].id = 96;
    SetSendParameters(parameters);
    const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
    EXPECT_EQ(96, spec.payload_type);
    EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
    SetSend(true);
    EXPECT_TRUE(channel_->CanInsertDtmf());
  }
  
  // Test that CanInsertDtmf() is governed by the send flag
  TEST_F(WebRtcVoiceEngineTestFake, DTMFControlledBySendFlag) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kTelephoneEventCodec1);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs[0].id = 98;  // DTMF
    parameters.codecs[1].id = 96;
    SetSendParameters(parameters);
    EXPECT_FALSE(channel_->CanInsertDtmf());
    SetSend(true);
    EXPECT_TRUE(channel_->CanInsertDtmf());
    SetSend(false);
    EXPECT_FALSE(channel_->CanInsertDtmf());
  }
  
  // Test that payload type range is limited for telephone-event codec.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kTelephoneEventCodec2);
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs[0].id = 0;  // DTMF
    parameters.codecs[1].id = 96;
    SetSendParameters(parameters);
    SetSend(true);
    EXPECT_TRUE(channel_->CanInsertDtmf());
    parameters.codecs[0].id = 128;  // DTMF
    EXPECT_FALSE(channel_->SetSendParameters(parameters));
    EXPECT_FALSE(channel_->CanInsertDtmf());
    parameters.codecs[0].id = 127;
    SetSendParameters(parameters);
    EXPECT_TRUE(channel_->CanInsertDtmf());
    parameters.codecs[0].id = -1;  // DTMF
    EXPECT_FALSE(channel_->SetSendParameters(parameters));
    EXPECT_FALSE(channel_->CanInsertDtmf());
  }
  
  // Test that we can set send codecs even with CN codec as the first
  // one on the list.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNOnTop) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kCn16000Codec);
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs[0].id = 98;  // wideband CN
    parameters.codecs[1].id = 96;
    SetSendParameters(parameters);
    const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
    EXPECT_EQ(96, send_codec_spec.payload_type);
    EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
    EXPECT_EQ(98, send_codec_spec.cng_payload_type);
  }
  
  // Test that we set VAD and DTMF types correctly as caller.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    // TODO(juberti): cn 32000
    parameters.codecs.push_back(kCn16000Codec);
    parameters.codecs.push_back(kCn8000Codec);
    parameters.codecs.push_back(kTelephoneEventCodec1);
    parameters.codecs[0].id = 96;
    parameters.codecs[2].id = 97;  // wideband CN
    parameters.codecs[4].id = 98;  // DTMF
    SetSendParameters(parameters);
    const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
    EXPECT_EQ(96, send_codec_spec.payload_type);
    EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
    EXPECT_EQ(1u, send_codec_spec.format.num_channels);
    EXPECT_EQ(97, send_codec_spec.cng_payload_type);
    SetSend(true);
    EXPECT_TRUE(channel_->CanInsertDtmf());
  }
  
  // Test that we set VAD and DTMF types correctly as callee.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
    EXPECT_TRUE(SetupChannel());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    // TODO(juberti): cn 32000
    parameters.codecs.push_back(kCn16000Codec);
    parameters.codecs.push_back(kCn8000Codec);
    parameters.codecs.push_back(kTelephoneEventCodec2);
    parameters.codecs[0].id = 96;
    parameters.codecs[2].id = 97;  // wideband CN
    parameters.codecs[4].id = 98;  // DTMF
    SetSendParameters(parameters);
    EXPECT_TRUE(
        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
  
    const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
    EXPECT_EQ(96, send_codec_spec.payload_type);
    EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
    EXPECT_EQ(1u, send_codec_spec.format.num_channels);
    EXPECT_EQ(97, send_codec_spec.cng_payload_type);
    SetSend(true);
    EXPECT_TRUE(channel_->CanInsertDtmf());
  }
  
  // Test that we only apply VAD if we have a CN codec that matches the
  // send codec clockrate.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    // Set ISAC(16K) and CN(16K). VAD should be activated.
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kCn16000Codec);
    parameters.codecs[1].id = 97;
    SetSendParameters(parameters);
    {
      const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
      EXPECT_EQ(1u, send_codec_spec.format.num_channels);
      EXPECT_EQ(97, send_codec_spec.cng_payload_type);
    }
    // Set PCMU(8K) and CN(16K). VAD should not be activated.
    parameters.codecs[0] = kPcmuCodec;
    SetSendParameters(parameters);
    {
      const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
      EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
    }
    // Set PCMU(8K) and CN(8K). VAD should be activated.
    parameters.codecs[1] = kCn8000Codec;
    SetSendParameters(parameters);
    {
      const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
      EXPECT_EQ(1u, send_codec_spec.format.num_channels);
      EXPECT_EQ(13, send_codec_spec.cng_payload_type);
    }
    // Set ISAC(16K) and CN(8K). VAD should not be activated.
    parameters.codecs[0] = kIsacCodec;
    SetSendParameters(parameters);
    {
      const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
      EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
      EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
    }
  }
  
  // Test that we perform case-insensitive matching of codec names.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioSendParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    parameters.codecs.push_back(kCn16000Codec);
    parameters.codecs.push_back(kCn8000Codec);
    parameters.codecs.push_back(kTelephoneEventCodec1);
    parameters.codecs[0].name = "iSaC";
    parameters.codecs[0].id = 96;
    parameters.codecs[2].id = 97;  // wideband CN
    parameters.codecs[4].id = 98;  // DTMF
    SetSendParameters(parameters);
    const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
    EXPECT_EQ(96, send_codec_spec.payload_type);
    EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
    EXPECT_EQ(1u, send_codec_spec.format.num_channels);
    EXPECT_EQ(97, send_codec_spec.cng_payload_type);
    SetSend(true);
    EXPECT_TRUE(channel_->CanInsertDtmf());
  }
  
  class WebRtcVoiceEngineWithSendSideBweTest : public WebRtcVoiceEngineTestFake {
   public:
    WebRtcVoiceEngineWithSendSideBweTest()
        : WebRtcVoiceEngineTestFake("WebRTC-Audio-SendSideBwe/Enabled/") {}
  };
  
  TEST_F(WebRtcVoiceEngineWithSendSideBweTest,
         SupportsTransportSequenceNumberHeaderExtension) {
    const cricket::RtpCapabilities capabilities = engine_->GetCapabilities();
    EXPECT_THAT(capabilities.header_extensions,
                Contains(::testing::Field(
                    "uri", &RtpExtension::uri,
                    webrtc::RtpExtension::kTransportSequenceNumberUri)));
  }
  
  // Test support for audio level header extension.
  TEST_F(WebRtcVoiceEngineTestFake, SendAudioLevelHeaderExtensions) {
    TestSetSendRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri);
  }
  TEST_F(WebRtcVoiceEngineTestFake, RecvAudioLevelHeaderExtensions) {
    TestSetRecvRtpHeaderExtensions(webrtc::RtpExtension::kAudioLevelUri);
  }
  
  // Test support for transport sequence number header extension.
  TEST_F(WebRtcVoiceEngineTestFake, SendTransportSequenceNumberHeaderExtensions) {
    TestSetSendRtpHeaderExtensions(
        webrtc::RtpExtension::kTransportSequenceNumberUri);
  }
  TEST_F(WebRtcVoiceEngineTestFake, RecvTransportSequenceNumberHeaderExtensions) {
    TestSetRecvRtpHeaderExtensions(
        webrtc::RtpExtension::kTransportSequenceNumberUri);
  }
  
  // Test that we can create a channel and start sending on it.
  TEST_F(WebRtcVoiceEngineTestFake, Send) {
    EXPECT_TRUE(SetupSendStream());
    SetSendParameters(send_parameters_);
    SetSend(true);
    EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
    SetSend(false);
    EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
  }
  
  // Test that a channel will send if and only if it has a source and is enabled
  // for sending.
  TEST_F(WebRtcVoiceEngineTestFake, SendStateWithAndWithoutSource) {
    EXPECT_TRUE(SetupSendStream());
    SetSendParameters(send_parameters_);
    SetAudioSend(kSsrcX, true, nullptr);
    SetSend(true);
    EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
    SetAudioSend(kSsrcX, true, &fake_source_);
    EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
    SetAudioSend(kSsrcX, true, nullptr);
    EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
  }
  
  // Test that a channel is muted/unmuted.
  TEST_F(WebRtcVoiceEngineTestFake, SendStateMuteUnmute) {
    EXPECT_TRUE(SetupSendStream());
    SetSendParameters(send_parameters_);
    EXPECT_FALSE(GetSendStream(kSsrcX).muted());
    SetAudioSend(kSsrcX, true, nullptr);
    EXPECT_FALSE(GetSendStream(kSsrcX).muted());
    SetAudioSend(kSsrcX, false, nullptr);
    EXPECT_TRUE(GetSendStream(kSsrcX).muted());
  }
  
  // Test that SetSendParameters() does not alter a stream's send state.
  TEST_F(WebRtcVoiceEngineTestFake, SendStateWhenStreamsAreRecreated) {
    EXPECT_TRUE(SetupSendStream());
    EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
  
    // Turn on sending.
    SetSend(true);
    EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
  
    // Changing RTP header extensions will recreate the AudioSendStream.
    send_parameters_.extensions.push_back(
        webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12));
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
  
    // Turn off sending.
    SetSend(false);
    EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
  
    // Changing RTP header extensions will recreate the AudioSendStream.
    send_parameters_.extensions.clear();
    SetSendParameters(send_parameters_);
    EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
  }
  
  // Test that we can create a channel and start playing out on it.
  TEST_F(WebRtcVoiceEngineTestFake, Playout) {
    EXPECT_TRUE(SetupRecvStream());
    EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
    channel_->SetPlayout(true);
    EXPECT_TRUE(GetRecvStream(kSsrcX).started());
    channel_->SetPlayout(false);
    EXPECT_FALSE(GetRecvStream(kSsrcX).started());
  }
  
  // Test that we can add and remove send streams.
  TEST_F(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) {
    SetupForMultiSendStream();
  
    // Set the global state for sending.
    SetSend(true);
  
    for (uint32_t ssrc : kSsrcs4) {
      EXPECT_TRUE(
          channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
      SetAudioSend(ssrc, true, &fake_source_);
      // Verify that we are in a sending state for all the created streams.
      EXPECT_TRUE(GetSendStream(ssrc).IsSending());
    }
    EXPECT_EQ(arraysize(kSsrcs4), call_.GetAudioSendStreams().size());
  
    // Delete the send streams.
    for (uint32_t ssrc : kSsrcs4) {
      EXPECT_TRUE(channel_->RemoveSendStream(ssrc));
      EXPECT_FALSE(call_.GetAudioSendStream(ssrc));
      EXPECT_FALSE(channel_->RemoveSendStream(ssrc));
    }
    EXPECT_EQ(0u, call_.GetAudioSendStreams().size());
  }
  
  // Test SetSendCodecs correctly configure the codecs in all send streams.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
    SetupForMultiSendStream();
  
    // Create send streams.
    for (uint32_t ssrc : kSsrcs4) {
      EXPECT_TRUE(
          channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
    }
  
    cricket::AudioSendParameters parameters;
    // Set ISAC(16K) and CN(16K). VAD should be activated.
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kCn16000Codec);
    parameters.codecs[1].id = 97;
    SetSendParameters(parameters);
  
    // Verify ISAC and VAD are corrected configured on all send channels.
    for (uint32_t ssrc : kSsrcs4) {
      ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr);
      const auto& send_codec_spec =
          *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
      EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
      EXPECT_EQ(1u, send_codec_spec.format.num_channels);
      EXPECT_EQ(97, send_codec_spec.cng_payload_type);
    }
  
    // Change to PCMU(8K) and CN(16K).
    parameters.codecs[0] = kPcmuCodec;
    SetSendParameters(parameters);
    for (uint32_t ssrc : kSsrcs4) {
      ASSERT_TRUE(call_.GetAudioSendStream(ssrc) != nullptr);
      const auto& send_codec_spec =
          *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
      EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
      EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
    }
  }
  
  // Test we can SetSend on all send streams correctly.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) {
    SetupForMultiSendStream();
  
    // Create the send channels and they should be a "not sending" date.
    for (uint32_t ssrc : kSsrcs4) {
      EXPECT_TRUE(
          channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
      SetAudioSend(ssrc, true, &fake_source_);
      EXPECT_FALSE(GetSendStream(ssrc).IsSending());
    }
  
    // Set the global state for starting sending.
    SetSend(true);
    for (uint32_t ssrc : kSsrcs4) {
      // Verify that we are in a sending state for all the send streams.
      EXPECT_TRUE(GetSendStream(ssrc).IsSending());
    }
  
    // Set the global state for stopping sending.
    SetSend(false);
    for (uint32_t ssrc : kSsrcs4) {
      // Verify that we are in a stop state for all the send streams.
      EXPECT_FALSE(GetSendStream(ssrc).IsSending());
    }
  }
  
  // Test we can set the correct statistics on all send streams.
  TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
    SetupForMultiSendStream();
  
    // Create send streams.
    for (uint32_t ssrc : kSsrcs4) {
      EXPECT_TRUE(
          channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
    }
  
    // Create a receive stream to check that none of the send streams end up in
    // the receive stream stats.
    EXPECT_TRUE(AddRecvStream(kSsrcY));
  
    // We need send codec to be set to get all stats.
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
    SetAudioSendStreamStats();
  
    // Check stats for the added streams.
    {
      cricket::VoiceMediaInfo info;
      EXPECT_EQ(true, channel_->GetStats(&info));
  
      // We have added 4 send streams. We should see empty stats for all.
      EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
      for (const auto& sender : info.senders) {
        VerifyVoiceSenderInfo(sender, false);
      }
      VerifyVoiceSendRecvCodecs(info);
  
      // We have added one receive stream. We should see empty stats.
      EXPECT_EQ(info.receivers.size(), 1u);
      EXPECT_EQ(info.receivers[0].ssrc(), 0u);
    }
  
    // Remove the kSsrcY stream. No receiver stats.
    {
      cricket::VoiceMediaInfo info;
      EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY));
      EXPECT_EQ(true, channel_->GetStats(&info));
      EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
      EXPECT_EQ(0u, info.receivers.size());
    }
  
    // Deliver a new packet - a default receive stream should be created and we
    // should see stats again.
    {
      cricket::VoiceMediaInfo info;
      DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
      SetAudioReceiveStreamStats();
      EXPECT_EQ(true, channel_->GetStats(&info));
      EXPECT_EQ(static_cast<size_t>(arraysize(kSsrcs4)), info.senders.size());
      EXPECT_EQ(1u, info.receivers.size());
      VerifyVoiceReceiverInfo(info.receivers[0]);
      VerifyVoiceSendRecvCodecs(info);
    }
  }
  
  // Test that we can add and remove receive streams, and do proper send/playout.
  // We can receive on multiple streams while sending one stream.
  TEST_F(WebRtcVoiceEngineTestFake, PlayoutWithMultipleStreams) {
    EXPECT_TRUE(SetupSendStream());
  
    // Start playout without a receive stream.
    SetSendParameters(send_parameters_);
    channel_->SetPlayout(true);
  
    // Adding another stream should enable playout on the new stream only.
    EXPECT_TRUE(AddRecvStream(kSsrcY));
    SetSend(true);
    EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
  
    // Make sure only the new stream is played out.
    EXPECT_TRUE(GetRecvStream(kSsrcY).started());
  
    // Adding yet another stream should have stream 2 and 3 enabled for playout.
    EXPECT_TRUE(AddRecvStream(kSsrcZ));
    EXPECT_TRUE(GetRecvStream(kSsrcY).started());
    EXPECT_TRUE(GetRecvStream(kSsrcZ).started());
  
    // Stop sending.
    SetSend(false);
    EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
  
    // Stop playout.
    channel_->SetPlayout(false);
    EXPECT_FALSE(GetRecvStream(kSsrcY).started());
    EXPECT_FALSE(GetRecvStream(kSsrcZ).started());
  
    // Restart playout and make sure recv streams are played out.
    channel_->SetPlayout(true);
    EXPECT_TRUE(GetRecvStream(kSsrcY).started());
    EXPECT_TRUE(GetRecvStream(kSsrcZ).started());
  
    // Now remove the recv streams.
    EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcZ));
    EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY));
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, TxAgcConfigViaOptions) {
    EXPECT_TRUE(SetupSendStream());
    EXPECT_CALL(adm_, BuiltInAGCIsAvailable())
        .Times(::testing::AtLeast(1))
        .WillRepeatedly(Return(false));
    const auto& agc_config = apm_config_.gain_controller1;
  
    // Ensure default options.
    VerifyGainControlEnabledCorrectly();
    VerifyGainControlDefaultSettings();
  
    send_parameters_.options.auto_gain_control = false;
    SetSendParameters(send_parameters_);
    EXPECT_FALSE(agc_config.enabled);
    send_parameters_.options.auto_gain_control = absl::nullopt;
  
    send_parameters_.options.tx_agc_target_dbov = 5;
    SetSendParameters(send_parameters_);
    EXPECT_EQ(5, agc_config.target_level_dbfs);
    send_parameters_.options.tx_agc_target_dbov = absl::nullopt;
  
    send_parameters_.options.tx_agc_digital_compression_gain = 10;
    SetSendParameters(send_parameters_);
    EXPECT_EQ(10, agc_config.compression_gain_db);
    send_parameters_.options.tx_agc_digital_compression_gain = absl::nullopt;
  
    send_parameters_.options.tx_agc_limiter = false;
    SetSendParameters(send_parameters_);
    EXPECT_FALSE(agc_config.enable_limiter);
    send_parameters_.options.tx_agc_limiter = absl::nullopt;
  
    SetSendParameters(send_parameters_);
    // Expect all options to have been preserved.
    EXPECT_FALSE(agc_config.enabled);
    EXPECT_EQ(5, agc_config.target_level_dbfs);
    EXPECT_EQ(10, agc_config.compression_gain_db);
    EXPECT_FALSE(agc_config.enable_limiter);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetAudioNetworkAdaptorViaOptions) {
    EXPECT_TRUE(SetupSendStream());
    send_parameters_.options.audio_network_adaptor = true;
    send_parameters_.options.audio_network_adaptor_config = {"1234"};
    SetSendParameters(send_parameters_);
    EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
              GetAudioNetworkAdaptorConfig(kSsrcX));
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, AudioSendResetAudioNetworkAdaptor) {
    EXPECT_TRUE(SetupSendStream());
    send_parameters_.options.audio_network_adaptor = true;
    send_parameters_.options.audio_network_adaptor_config = {"1234"};
    SetSendParameters(send_parameters_);
    EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
              GetAudioNetworkAdaptorConfig(kSsrcX));
    cricket::AudioOptions options;
    options.audio_network_adaptor = false;
    SetAudioSend(kSsrcX, true, nullptr, &options);
    EXPECT_EQ(absl::nullopt, GetAudioNetworkAdaptorConfig(kSsrcX));
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) {
    EXPECT_TRUE(SetupSendStream());
    send_parameters_.options.audio_network_adaptor = true;
    send_parameters_.options.audio_network_adaptor_config = {"1234"};
    SetSendParameters(send_parameters_);
    EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
              GetAudioNetworkAdaptorConfig(kSsrcX));
    const int initial_num = call_.GetNumCreatedSendStreams();
    cricket::AudioOptions options;
    options.audio_network_adaptor = absl::nullopt;
    // Unvalued |options.audio_network_adaptor|.should not reset audio network
    // adaptor.
    SetAudioSend(kSsrcX, true, nullptr, &options);
    // AudioSendStream not expected to be recreated.
    EXPECT_EQ(initial_num, call_.GetNumCreatedSendStreams());
    EXPECT_EQ(send_parameters_.options.audio_network_adaptor_config,
              GetAudioNetworkAdaptorConfig(kSsrcX));
  }
  
  class WebRtcVoiceEngineWithSendSideBweWithOverheadTest
      : public WebRtcVoiceEngineTestFake {
   public:
    WebRtcVoiceEngineWithSendSideBweWithOverheadTest()
        : WebRtcVoiceEngineTestFake(
              "WebRTC-Audio-SendSideBwe/Enabled/WebRTC-Audio-Allocation/"
              "min:6000bps,max:32000bps/WebRTC-SendSideBwe-WithOverhead/"
              "Enabled/") {}
  };
  
  // Test that we can set the outgoing SSRC properly.
  // SSRC is set in SetupSendStream() by calling AddSendStream.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrc) {
    EXPECT_TRUE(SetupSendStream());
    EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, GetStats) {
    // Setup. We need send codec to be set to get all stats.
    EXPECT_TRUE(SetupSendStream());
    // SetupSendStream adds a send stream with kSsrcX, so the receive
    // stream has to use a different SSRC.
    EXPECT_TRUE(AddRecvStream(kSsrcY));
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(channel_->SetRecvParameters(recv_parameters_));
    SetAudioSendStreamStats();
  
    // Check stats for the added streams.
    {
      cricket::VoiceMediaInfo info;
      EXPECT_EQ(true, channel_->GetStats(&info));
  
      // We have added one send stream. We should see the stats we've set.
      EXPECT_EQ(1u, info.senders.size());
      VerifyVoiceSenderInfo(info.senders[0], false);
      // We have added one receive stream. We should see empty stats.
      EXPECT_EQ(info.receivers.size(), 1u);
      EXPECT_EQ(info.receivers[0].ssrc(), 0u);
    }
  
    // Start sending - this affects some reported stats.
    {
      cricket::VoiceMediaInfo info;
      SetSend(true);
      EXPECT_EQ(true, channel_->GetStats(&info));
      VerifyVoiceSenderInfo(info.senders[0], true);
      VerifyVoiceSendRecvCodecs(info);
    }
  
    // Remove the kSsrcY stream. No receiver stats.
    {
      cricket::VoiceMediaInfo info;
      EXPECT_TRUE(channel_->RemoveRecvStream(kSsrcY));
      EXPECT_EQ(true, channel_->GetStats(&info));
      EXPECT_EQ(1u, info.senders.size());
      EXPECT_EQ(0u, info.receivers.size());
    }
  
    // Deliver a new packet - a default receive stream should be created and we
    // should see stats again.
    {
      cricket::VoiceMediaInfo info;
      DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
      SetAudioReceiveStreamStats();
      EXPECT_EQ(true, channel_->GetStats(&info));
      EXPECT_EQ(1u, info.senders.size());
      EXPECT_EQ(1u, info.receivers.size());
      VerifyVoiceReceiverInfo(info.receivers[0]);
      VerifyVoiceSendRecvCodecs(info);
    }
  }
  
  // Test that we can set the outgoing SSRC properly with multiple streams.
  // SSRC is set in SetupSendStream() by calling AddSendStream.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcWithMultipleStreams) {
    EXPECT_TRUE(SetupSendStream());
    EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
    EXPECT_TRUE(AddRecvStream(kSsrcY));
    EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
  }
  
  // Test that the local SSRC is the same on sending and receiving channels if the
  // receive channel is created before the send channel.
  TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) {
    EXPECT_TRUE(SetupChannel());
    EXPECT_TRUE(AddRecvStream(kSsrcY));
    EXPECT_TRUE(
        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
    EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
    EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
  }
  
  // Test that we can properly receive packets.
  TEST_F(WebRtcVoiceEngineTestFake, Recv) {
    EXPECT_TRUE(SetupChannel());
    EXPECT_TRUE(AddRecvStream(1));
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
  
    EXPECT_TRUE(
        GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
  }
  
  // Test that we can properly receive packets on multiple streams.
  TEST_F(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) {
    EXPECT_TRUE(SetupChannel());
    const uint32_t ssrc1 = 1;
    const uint32_t ssrc2 = 2;
    const uint32_t ssrc3 = 3;
    EXPECT_TRUE(AddRecvStream(ssrc1));
    EXPECT_TRUE(AddRecvStream(ssrc2));
    EXPECT_TRUE(AddRecvStream(ssrc3));
    // Create packets with the right SSRCs.
    unsigned char packets[4][sizeof(kPcmuFrame)];
    for (size_t i = 0; i < arraysize(packets); ++i) {
      memcpy(packets[i], kPcmuFrame, sizeof(kPcmuFrame));
      rtc::SetBE32(packets[i] + 8, static_cast<uint32_t>(i));
    }
  
    const cricket::FakeAudioReceiveStream& s1 = GetRecvStream(ssrc1);
    const cricket::FakeAudioReceiveStream& s2 = GetRecvStream(ssrc2);
    const cricket::FakeAudioReceiveStream& s3 = GetRecvStream(ssrc3);
  
    EXPECT_EQ(s1.received_packets(), 0);
    EXPECT_EQ(s2.received_packets(), 0);
    EXPECT_EQ(s3.received_packets(), 0);
  
    DeliverPacket(packets[0], sizeof(packets[0]));
    EXPECT_EQ(s1.received_packets(), 0);
    EXPECT_EQ(s2.received_packets(), 0);
    EXPECT_EQ(s3.received_packets(), 0);
  
    DeliverPacket(packets[1], sizeof(packets[1]));
    EXPECT_EQ(s1.received_packets(), 1);
    EXPECT_TRUE(s1.VerifyLastPacket(packets[1], sizeof(packets[1])));
    EXPECT_EQ(s2.received_packets(), 0);
    EXPECT_EQ(s3.received_packets(), 0);
  
    DeliverPacket(packets[2], sizeof(packets[2]));
    EXPECT_EQ(s1.received_packets(), 1);
    EXPECT_EQ(s2.received_packets(), 1);
    EXPECT_TRUE(s2.VerifyLastPacket(packets[2], sizeof(packets[2])));
    EXPECT_EQ(s3.received_packets(), 0);
  
    DeliverPacket(packets[3], sizeof(packets[3]));
    EXPECT_EQ(s1.received_packets(), 1);
    EXPECT_EQ(s2.received_packets(), 1);
    EXPECT_EQ(s3.received_packets(), 1);
    EXPECT_TRUE(s3.VerifyLastPacket(packets[3], sizeof(packets[3])));
  
    EXPECT_TRUE(channel_->RemoveRecvStream(ssrc3));
    EXPECT_TRUE(channel_->RemoveRecvStream(ssrc2));
    EXPECT_TRUE(channel_->RemoveRecvStream(ssrc1));
  }
  
  // Test that receiving on an unsignaled stream works (a stream is created).
  TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaled) {
    EXPECT_TRUE(SetupChannel());
    EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
  
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
  
    EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
    EXPECT_TRUE(
        GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
  }
  
  // Tests that when we add a stream without SSRCs, but contains a stream_id
  // that it is stored and its stream id is later used when the first packet
  // arrives to properly create a receive stream with a sync label.
  TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaledSsrcWithSignaledStreamId) {
    const char kSyncLabel[] = "sync_label";
    EXPECT_TRUE(SetupChannel());
    cricket::StreamParams unsignaled_stream;
    unsignaled_stream.set_stream_ids({kSyncLabel});
    ASSERT_TRUE(channel_->AddRecvStream(unsignaled_stream));
    // The stream shouldn't have been created at this point because it doesn't
    // have any SSRCs.
    EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
  
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
  
    EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
    EXPECT_TRUE(
        GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
    EXPECT_EQ(kSyncLabel, GetRecvStream(kSsrc1).GetConfig().sync_group);
  
    // Removing the unsignaled stream clears the cached parameters. If a new
    // default unsignaled receive stream is created it will not have a sync group.
    channel_->RemoveRecvStream(0);
    channel_->RemoveRecvStream(kSsrc1);
  
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
  
    EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
    EXPECT_TRUE(
        GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
    EXPECT_TRUE(GetRecvStream(kSsrc1).GetConfig().sync_group.empty());
  }
  
  // Test that receiving N unsignaled stream works (streams will be created), and
  // that packets are forwarded to them all.
  TEST_F(WebRtcVoiceEngineTestFake, RecvMultipleUnsignaled) {
    EXPECT_TRUE(SetupChannel());
    unsigned char packet[sizeof(kPcmuFrame)];
    memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
  
    // Note that SSRC = 0 is not supported.
    for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) {
      rtc::SetBE32(&packet[8], ssrc);
      DeliverPacket(packet, sizeof(packet));
  
      // Verify we have one new stream for each loop iteration.
      EXPECT_EQ(ssrc, call_.GetAudioReceiveStreams().size());
      EXPECT_EQ(1, GetRecvStream(ssrc).received_packets());
      EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet)));
    }
  
    // Sending on the same SSRCs again should not create new streams.
    for (uint32_t ssrc = 1; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc) {
      rtc::SetBE32(&packet[8], ssrc);
      DeliverPacket(packet, sizeof(packet));
  
      EXPECT_EQ(kMaxUnsignaledRecvStreams, call_.GetAudioReceiveStreams().size());
      EXPECT_EQ(2, GetRecvStream(ssrc).received_packets());
      EXPECT_TRUE(GetRecvStream(ssrc).VerifyLastPacket(packet, sizeof(packet)));
    }
  
    // Send on another SSRC, the oldest unsignaled stream (SSRC=1) is replaced.
    constexpr uint32_t kAnotherSsrc = 667;
    rtc::SetBE32(&packet[8], kAnotherSsrc);
    DeliverPacket(packet, sizeof(packet));
  
    const auto& streams = call_.GetAudioReceiveStreams();
    EXPECT_EQ(kMaxUnsignaledRecvStreams, streams.size());
    size_t i = 0;
    for (uint32_t ssrc = 2; ssrc < (1 + kMaxUnsignaledRecvStreams); ++ssrc, ++i) {
      EXPECT_EQ(ssrc, streams[i]->GetConfig().rtp.remote_ssrc);
      EXPECT_EQ(2, streams[i]->received_packets());
    }
    EXPECT_EQ(kAnotherSsrc, streams[i]->GetConfig().rtp.remote_ssrc);
    EXPECT_EQ(1, streams[i]->received_packets());
    // Sanity check that we've checked all streams.
    EXPECT_EQ(kMaxUnsignaledRecvStreams, (i + 1));
  }
  
  // Test that a default channel is created even after a signaled stream has been
  // added, and that this stream will get any packets for unknown SSRCs.
  TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
    EXPECT_TRUE(SetupChannel());
    unsigned char packet[sizeof(kPcmuFrame)];
    memcpy(packet, kPcmuFrame, sizeof(kPcmuFrame));
  
    // Add a known stream, send packet and verify we got it.
    const uint32_t signaled_ssrc = 1;
    rtc::SetBE32(&packet[8], signaled_ssrc);
    EXPECT_TRUE(AddRecvStream(signaled_ssrc));
    DeliverPacket(packet, sizeof(packet));
    EXPECT_TRUE(
        GetRecvStream(signaled_ssrc).VerifyLastPacket(packet, sizeof(packet)));
    EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
  
    // Note that the first unknown SSRC cannot be 0, because we only support
    // creating receive streams for SSRC!=0.
    const uint32_t unsignaled_ssrc = 7011;
    rtc::SetBE32(&packet[8], unsignaled_ssrc);
    DeliverPacket(packet, sizeof(packet));
    EXPECT_TRUE(
        GetRecvStream(unsignaled_ssrc).VerifyLastPacket(packet, sizeof(packet)));
    EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
  
    DeliverPacket(packet, sizeof(packet));
    EXPECT_EQ(2, GetRecvStream(unsignaled_ssrc).received_packets());
  
    rtc::SetBE32(&packet[8], signaled_ssrc);
    DeliverPacket(packet, sizeof(packet));
    EXPECT_EQ(2, GetRecvStream(signaled_ssrc).received_packets());
    EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
  }
  
  // Two tests to verify that adding a receive stream with the same SSRC as a
  // previously added unsignaled stream will only recreate underlying stream
  // objects if the stream parameters have changed.
  TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_NoRecreate) {
    EXPECT_TRUE(SetupChannel());
  
    // Spawn unsignaled stream with SSRC=1.
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
    EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
    EXPECT_TRUE(
        GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
  
    // Verify that the underlying stream object in Call is not recreated when a
    // stream with SSRC=1 is added.
    const auto& streams = call_.GetAudioReceiveStreams();
    EXPECT_EQ(1u, streams.size());
    int audio_receive_stream_id = streams.front()->id();
    EXPECT_TRUE(AddRecvStream(1));
    EXPECT_EQ(1u, streams.size());
    EXPECT_EQ(audio_receive_stream_id, streams.front()->id());
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Recreate) {
    EXPECT_TRUE(SetupChannel());
  
    // Spawn unsignaled stream with SSRC=1.
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
    EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
    EXPECT_TRUE(
        GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
  
    // Verify that the underlying stream object in Call *is* recreated when a
    // stream with SSRC=1 is added, and which has changed stream parameters.
    const auto& streams = call_.GetAudioReceiveStreams();
    EXPECT_EQ(1u, streams.size());
    int audio_receive_stream_id = streams.front()->id();
    cricket::StreamParams stream_params;
    stream_params.ssrcs.push_back(1);
    stream_params.set_stream_ids({"stream_id"});
    EXPECT_TRUE(channel_->AddRecvStream(stream_params));
    EXPECT_EQ(1u, streams.size());
    EXPECT_NE(audio_receive_stream_id, streams.front()->id());
  }
  
  // Test that AddRecvStream creates new stream.
  TEST_F(WebRtcVoiceEngineTestFake, AddRecvStream) {
    EXPECT_TRUE(SetupRecvStream());
    EXPECT_TRUE(AddRecvStream(1));
  }
  
  // Test that after adding a recv stream, we do not decode more codecs than
  // those previously passed into SetRecvCodecs.
  TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamUnsupportedCodec) {
    EXPECT_TRUE(SetupSendStream());
    cricket::AudioRecvParameters parameters;
    parameters.codecs.push_back(kIsacCodec);
    parameters.codecs.push_back(kPcmuCodec);
    EXPECT_TRUE(channel_->SetRecvParameters(parameters));
    EXPECT_TRUE(AddRecvStream(kSsrcX));
    EXPECT_THAT(GetRecvStreamConfig(kSsrcX).decoder_map,
                (ContainerEq<std::map<int, webrtc::SdpAudioFormat>>(
                    {{0, {"PCMU", 8000, 1}}, {103, {"ISAC", 16000, 1}}})));
  }
  
  // Test that we properly clean up any streams that were added, even if
  // not explicitly removed.
  TEST_F(WebRtcVoiceEngineTestFake, StreamCleanup) {
    EXPECT_TRUE(SetupSendStream());
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(AddRecvStream(1));
    EXPECT_TRUE(AddRecvStream(2));
  
    EXPECT_EQ(1u, call_.GetAudioSendStreams().size());
    EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
    delete channel_;
    channel_ = NULL;
    EXPECT_EQ(0u, call_.GetAudioSendStreams().size());
    EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithZeroSsrc) {
    EXPECT_TRUE(SetupSendStream());
    EXPECT_FALSE(AddRecvStream(0));
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithSameSsrc) {
    EXPECT_TRUE(SetupChannel());
    EXPECT_TRUE(AddRecvStream(1));
    EXPECT_FALSE(AddRecvStream(1));
  }
  
  // Test the InsertDtmf on default send stream as caller.
  TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCaller) {
    TestInsertDtmf(0, true, kTelephoneEventCodec1);
  }
  
  // Test the InsertDtmf on default send stream as callee
  TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnDefaultSendStreamAsCallee) {
    TestInsertDtmf(0, false, kTelephoneEventCodec2);
  }
  
  // Test the InsertDtmf on specified send stream as caller.
  TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCaller) {
    TestInsertDtmf(kSsrcX, true, kTelephoneEventCodec2);
  }
  
  // Test the InsertDtmf on specified send stream as callee.
  TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) {
    TestInsertDtmf(kSsrcX, false, kTelephoneEventCodec1);
  }
  
  // Test propagation of extmap allow mixed setting.
  TEST_F(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedAsCaller) {
    TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/true);
  }
  TEST_F(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedDisabledAsCaller) {
    TestExtmapAllowMixedCaller(/*extmap_allow_mixed=*/false);
  }
  TEST_F(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedAsCallee) {
    TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/true);
  }
  TEST_F(WebRtcVoiceEngineTestFake, SetExtmapAllowMixedDisabledAsCallee) {
    TestExtmapAllowMixedCallee(/*extmap_allow_mixed=*/false);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
    EXPECT_TRUE(SetupSendStream());
    EXPECT_TRUE(AddRecvStream(kSsrcY));
    EXPECT_CALL(adm_, BuiltInAECIsAvailable())
        .Times(9)
        .WillRepeatedly(Return(false));
    EXPECT_CALL(adm_, BuiltInAGCIsAvailable())
        .Times(4)
        .WillRepeatedly(Return(false));
    EXPECT_CALL(adm_, BuiltInNSIsAvailable())
        .Times(2)
        .WillRepeatedly(Return(false));
  
    EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
    EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate);
  
    // Nothing set in AudioOptions, so everything should be as default.
    send_parameters_.options = cricket::AudioOptions();
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(IsEchoCancellationEnabled());
    EXPECT_TRUE(IsHighPassFilterEnabled());
    EXPECT_TRUE(IsTypingDetectionEnabled());
    EXPECT_EQ(200u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
    EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate);
  
    // Turn typing detection off.
    send_parameters_.options.typing_detection = false;
    SetSendParameters(send_parameters_);
    EXPECT_FALSE(IsTypingDetectionEnabled());
  
    // Leave typing detection unchanged, but non-default.
    send_parameters_.options.typing_detection = absl::nullopt;
    SetSendParameters(send_parameters_);
    EXPECT_FALSE(IsTypingDetectionEnabled());
  
    // Turn typing detection on.
    send_parameters_.options.typing_detection = true;
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(IsTypingDetectionEnabled());
  
    // Turn echo cancellation off
    send_parameters_.options.echo_cancellation = false;
    SetSendParameters(send_parameters_);
    EXPECT_FALSE(IsEchoCancellationEnabled());
  
    // Turn echo cancellation back on, with settings, and make sure
    // nothing else changed.
    send_parameters_.options.echo_cancellation = true;
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(IsEchoCancellationEnabled());
  
    // Turn on delay agnostic aec and make sure nothing change w.r.t. echo
    // control.
    send_parameters_.options.delay_agnostic_aec = true;
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(IsEchoCancellationEnabled());
  
    // Turn off echo cancellation and delay agnostic aec.
    send_parameters_.options.delay_agnostic_aec = false;
    send_parameters_.options.extended_filter_aec = false;
    send_parameters_.options.echo_cancellation = false;
    SetSendParameters(send_parameters_);
    EXPECT_FALSE(IsEchoCancellationEnabled());
  
    // Turning delay agnostic aec back on should also turn on echo cancellation.
    send_parameters_.options.delay_agnostic_aec = true;
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(IsEchoCancellationEnabled());
  
    // Turn off AGC
    send_parameters_.options.auto_gain_control = false;
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(IsEchoCancellationEnabled());
    EXPECT_FALSE(apm_config_.gain_controller1.enabled);
  
    // Turn AGC back on
    send_parameters_.options.auto_gain_control = true;
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(IsEchoCancellationEnabled());
    EXPECT_TRUE(apm_config_.gain_controller1.enabled);
  
    // Turn off other options.
    EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
    EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
    send_parameters_.options.noise_suppression = false;
    send_parameters_.options.highpass_filter = false;
    send_parameters_.options.stereo_swapping = true;
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(IsEchoCancellationEnabled());
    EXPECT_FALSE(IsHighPassFilterEnabled());
    EXPECT_TRUE(apm_config_.gain_controller1.enabled);
  
    // Set options again to ensure it has no impact.
    EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
    EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(IsEchoCancellationEnabled());
    EXPECT_TRUE(apm_config_.gain_controller1.enabled);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
    EXPECT_TRUE(SetupSendStream());
    EXPECT_CALL(adm_, BuiltInAECIsAvailable())
        .Times(8)
        .WillRepeatedly(Return(false));
    EXPECT_CALL(adm_, BuiltInAGCIsAvailable())
        .Times(8)
        .WillRepeatedly(Return(false));
    EXPECT_CALL(adm_, BuiltInNSIsAvailable())
        .Times(8)
        .WillRepeatedly(Return(false));
    EXPECT_CALL(adm_, RecordingIsInitialized())
        .Times(2)
        .WillRepeatedly(Return(false));
    EXPECT_CALL(adm_, Recording()).Times(2).WillRepeatedly(Return(false));
    EXPECT_CALL(adm_, InitRecording()).Times(2).WillRepeatedly(Return(0));
    EXPECT_CALL(*apm_, SetExtraOptions(::testing::_)).Times(10);
  
    std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel1(
        static_cast<cricket::WebRtcVoiceMediaChannel*>(
            engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
                                        cricket::AudioOptions(),
                                        webrtc::CryptoOptions())));
    std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel2(
        static_cast<cricket::WebRtcVoiceMediaChannel*>(
            engine_->CreateMediaChannel(&call_, cricket::MediaConfig(),
                                        cricket::AudioOptions(),
                                        webrtc::CryptoOptions())));
  
    // Have to add a stream to make SetSend work.
    cricket::StreamParams stream1;
    stream1.ssrcs.push_back(1);
    channel1->AddSendStream(stream1);
    cricket::StreamParams stream2;
    stream2.ssrcs.push_back(2);
    channel2->AddSendStream(stream2);
  
    // AEC and AGC and NS
    cricket::AudioSendParameters parameters_options_all = send_parameters_;
    parameters_options_all.options.echo_cancellation = true;
    parameters_options_all.options.auto_gain_control = true;
    parameters_options_all.options.noise_suppression = true;
    EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).Times(2).WillOnce(Return(0));
    EXPECT_CALL(apm_ns_, Enable(true)).Times(2).WillRepeatedly(Return(0));
    EXPECT_TRUE(channel1->SetSendParameters(parameters_options_all));
    EXPECT_TRUE(IsEchoCancellationEnabled());
    VerifyGainControlEnabledCorrectly();
    EXPECT_EQ(parameters_options_all.options, channel1->options());
    EXPECT_TRUE(channel2->SetSendParameters(parameters_options_all));
    EXPECT_TRUE(IsEchoCancellationEnabled());
    VerifyGainControlEnabledCorrectly();
    EXPECT_EQ(parameters_options_all.options, channel2->options());
  
    // unset NS
    cricket::AudioSendParameters parameters_options_no_ns = send_parameters_;
    parameters_options_no_ns.options.noise_suppression = false;
    EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
    EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
    EXPECT_TRUE(channel1->SetSendParameters(parameters_options_no_ns));
    EXPECT_TRUE(IsEchoCancellationEnabled());
    VerifyGainControlEnabledCorrectly();
    cricket::AudioOptions expected_options = parameters_options_all.options;
    expected_options.echo_cancellation = true;
    expected_options.auto_gain_control = true;
    expected_options.noise_suppression = false;
    EXPECT_EQ(expected_options, channel1->options());
  
    // unset AGC
    cricket::AudioSendParameters parameters_options_no_agc = send_parameters_;
    parameters_options_no_agc.options.auto_gain_control = false;
    EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
    EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0));
    EXPECT_TRUE(channel2->SetSendParameters(parameters_options_no_agc));
    EXPECT_TRUE(IsEchoCancellationEnabled());
    EXPECT_FALSE(apm_config_.gain_controller1.enabled);
    expected_options.echo_cancellation = true;
    expected_options.auto_gain_control = false;
    expected_options.noise_suppression = true;
    EXPECT_EQ(expected_options, channel2->options());
  
    EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
    EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0));
    EXPECT_TRUE(channel_->SetSendParameters(parameters_options_all));
    EXPECT_TRUE(IsEchoCancellationEnabled());
    VerifyGainControlEnabledCorrectly();
  
    EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
    EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
    channel1->SetSend(true);
    EXPECT_TRUE(IsEchoCancellationEnabled());
    VerifyGainControlEnabledCorrectly();
  
    EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
    EXPECT_CALL(apm_ns_, Enable(true)).WillOnce(Return(0));
    channel2->SetSend(true);
    EXPECT_TRUE(IsEchoCancellationEnabled());
    EXPECT_FALSE(apm_config_.gain_controller1.enabled);
  
    // Make sure settings take effect while we are sending.
    cricket::AudioSendParameters parameters_options_no_agc_nor_ns =
        send_parameters_;
    parameters_options_no_agc_nor_ns.options.auto_gain_control = false;
    parameters_options_no_agc_nor_ns.options.noise_suppression = false;
    EXPECT_CALL(apm_ns_, set_level(kDefaultNsLevel)).WillOnce(Return(0));
    EXPECT_CALL(apm_ns_, Enable(false)).WillOnce(Return(0));
    EXPECT_TRUE(channel2->SetSendParameters(parameters_options_no_agc_nor_ns));
    EXPECT_TRUE(IsEchoCancellationEnabled());
    EXPECT_FALSE(apm_config_.gain_controller1.enabled);
    expected_options.echo_cancellation = true;
    expected_options.auto_gain_control = false;
    expected_options.noise_suppression = false;
    EXPECT_EQ(expected_options, channel2->options());
  }
  
  // This test verifies DSCP settings are properly applied on voice media channel.
  TEST_F(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
    EXPECT_TRUE(SetupSendStream());
    cricket::FakeNetworkInterface network_interface;
    cricket::MediaConfig config;
    std::unique_ptr<cricket::WebRtcVoiceMediaChannel> channel;
    webrtc::RtpParameters parameters;
  
    EXPECT_CALL(*apm_, SetExtraOptions(::testing::_)).Times(3);
  
    channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
        engine_->CreateMediaChannel(&call_, config, cricket::AudioOptions(),
                                    webrtc::CryptoOptions())));
    channel->SetInterface(&network_interface, webrtc::MediaTransportConfig());
    // Default value when DSCP is disabled should be DSCP_DEFAULT.
    EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
  
    config.enable_dscp = true;
    channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
        engine_->CreateMediaChannel(&call_, config, cricket::AudioOptions(),
                                    webrtc::CryptoOptions())));
    channel->SetInterface(&network_interface, webrtc::MediaTransportConfig());
    EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
  
    // Create a send stream to configure
    EXPECT_TRUE(
        channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcZ)));
    parameters = channel->GetRtpSendParameters(kSsrcZ);
    ASSERT_FALSE(parameters.encodings.empty());
  
    // Various priorities map to various dscp values.
    parameters.encodings[0].network_priority = 4.0;
    ASSERT_TRUE(channel->SetRtpSendParameters(kSsrcZ, parameters).ok());
    EXPECT_EQ(rtc::DSCP_EF, network_interface.dscp());
    parameters.encodings[0].network_priority = 0.5;
    ASSERT_TRUE(channel->SetRtpSendParameters(kSsrcZ, parameters).ok());
    EXPECT_EQ(rtc::DSCP_CS1, network_interface.dscp());
  
    // A bad priority does not change the dscp value.
    parameters.encodings[0].network_priority = 0.0;
    ASSERT_FALSE(channel->SetRtpSendParameters(kSsrcZ, parameters).ok());
    EXPECT_EQ(rtc::DSCP_CS1, network_interface.dscp());
  
    // Packets should also self-identify their dscp in PacketOptions.
    const uint8_t kData[10] = {0};
    EXPECT_TRUE(channel->SendRtcp(kData, sizeof(kData)));
    EXPECT_EQ(rtc::DSCP_CS1, network_interface.options().dscp);
  
    // Verify that setting the option to false resets the
    // DiffServCodePoint.
    config.enable_dscp = false;
    channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
        engine_->CreateMediaChannel(&call_, config, cricket::AudioOptions(),
                                    webrtc::CryptoOptions())));
    channel->SetInterface(&network_interface, webrtc::MediaTransportConfig());
    // Default value when DSCP is disabled should be DSCP_DEFAULT.
    EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
  
    channel->SetInterface(nullptr, webrtc::MediaTransportConfig());
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolume) {
    EXPECT_TRUE(SetupChannel());
    EXPECT_FALSE(channel_->SetOutputVolume(kSsrcY, 0.5));
    cricket::StreamParams stream;
    stream.ssrcs.push_back(kSsrcY);
    EXPECT_TRUE(channel_->AddRecvStream(stream));
    EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrcY).gain());
    EXPECT_TRUE(channel_->SetOutputVolume(kSsrcY, 3));
    EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcY).gain());
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) {
    EXPECT_TRUE(SetupChannel());
  
    // Spawn an unsignaled stream by sending a packet - gain should be 1.
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
    EXPECT_DOUBLE_EQ(1, GetRecvStream(kSsrc1).gain());
  
    // Should remember the volume "2" which will be set on new unsignaled streams,
    // and also set the gain to 2 on existing unsignaled streams.
    EXPECT_TRUE(channel_->SetOutputVolume(kSsrc0, 2));
    EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrc1).gain());
  
    // Spawn an unsignaled stream by sending a packet - gain should be 2.
    unsigned char pcmuFrame2[sizeof(kPcmuFrame)];
    memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
    rtc::SetBE32(&pcmuFrame2[8], kSsrcX);
    DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
    EXPECT_DOUBLE_EQ(2, GetRecvStream(kSsrcX).gain());
  
    // Setting gain with SSRC=0 should affect all unsignaled streams.
    EXPECT_TRUE(channel_->SetOutputVolume(kSsrc0, 3));
    if (kMaxUnsignaledRecvStreams > 1) {
      EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain());
    }
    EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrcX).gain());
  
    // Setting gain on an individual stream affects only that.
    EXPECT_TRUE(channel_->SetOutputVolume(kSsrcX, 4));
    if (kMaxUnsignaledRecvStreams > 1) {
      EXPECT_DOUBLE_EQ(3, GetRecvStream(kSsrc1).gain());
    }
    EXPECT_DOUBLE_EQ(4, GetRecvStream(kSsrcX).gain());
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, BaseMinimumPlayoutDelayMs) {
    EXPECT_TRUE(SetupChannel());
    EXPECT_FALSE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrcY, 200));
    EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
  
    cricket::StreamParams stream;
    stream.ssrcs.push_back(kSsrcY);
    EXPECT_TRUE(channel_->AddRecvStream(stream));
    EXPECT_EQ(0, GetRecvStream(kSsrcY).base_mininum_playout_delay_ms());
    EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrcY, 300));
    EXPECT_EQ(300, GetRecvStream(kSsrcY).base_mininum_playout_delay_ms());
  }
  
  TEST_F(WebRtcVoiceEngineTestFake,
         BaseMinimumPlayoutDelayMsUnsignaledRecvStream) {
    // Here base minimum delay is abbreviated to delay in comments for shortness.
    EXPECT_TRUE(SetupChannel());
  
    // Spawn an unsignaled stream by sending a packet - delay should be 0.
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
    EXPECT_EQ(0, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1));
    // Check that it doesn't provide default values for unknown ssrc.
    EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
  
    // Check that default value for unsignaled streams is 0.
    EXPECT_EQ(0, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1));
  
    // Should remember the delay 100 which will be set on new unsignaled streams,
    // and also set the delay to 100 on existing unsignaled streams.
    EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrc0, 100));
    EXPECT_EQ(100, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1));
    // Check that it doesn't provide default values for unknown ssrc.
    EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
  
    // Spawn an unsignaled stream by sending a packet - delay should be 100.
    unsigned char pcmuFrame2[sizeof(kPcmuFrame)];
    memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
    rtc::SetBE32(&pcmuFrame2[8], kSsrcX);
    DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
    EXPECT_EQ(100, channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
  
    // Setting delay with SSRC=0 should affect all unsignaled streams.
    EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrc0, 300));
    if (kMaxUnsignaledRecvStreams > 1) {
      EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1));
    }
    EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
  
    // Setting delay on an individual stream affects only that.
    EXPECT_TRUE(channel_->SetBaseMinimumPlayoutDelayMs(kSsrcX, 400));
    if (kMaxUnsignaledRecvStreams > 1) {
      EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc1).value_or(-1));
    }
    EXPECT_EQ(400, channel_->GetBaseMinimumPlayoutDelayMs(kSsrcX).value_or(-1));
    EXPECT_EQ(300, channel_->GetBaseMinimumPlayoutDelayMs(kSsrc0).value_or(-1));
    // Check that it doesn't provide default values for unknown ssrc.
    EXPECT_FALSE(channel_->GetBaseMinimumPlayoutDelayMs(kSsrcY).has_value());
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetsSyncGroupFromStreamId) {
    const uint32_t kAudioSsrc = 123;
    const std::string kStreamId = "AvSyncLabel";
  
    EXPECT_TRUE(SetupSendStream());
    cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kAudioSsrc);
    sp.set_stream_ids({kStreamId});
    // Creating two channels to make sure that sync label is set properly for both
    // the default voice channel and following ones.
    EXPECT_TRUE(channel_->AddRecvStream(sp));
    sp.ssrcs[0] += 1;
    EXPECT_TRUE(channel_->AddRecvStream(sp));
  
    ASSERT_EQ(2u, call_.GetAudioReceiveStreams().size());
    EXPECT_EQ(kStreamId,
              call_.GetAudioReceiveStream(kAudioSsrc)->GetConfig().sync_group)
        << "SyncGroup should be set based on stream id";
    EXPECT_EQ(kStreamId,
              call_.GetAudioReceiveStream(kAudioSsrc + 1)->GetConfig().sync_group)
        << "SyncGroup should be set based on stream id";
  }
  
  // TODO(solenberg): Remove, once recv streams are configured through Call.
  //                  (This is then covered by TestSetRecvRtpHeaderExtensions.)
  TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
    // Test that setting the header extensions results in the expected state
    // changes on an associated Call.
    std::vector<uint32_t> ssrcs;
    ssrcs.push_back(223);
    ssrcs.push_back(224);
  
    EXPECT_TRUE(SetupSendStream());
    SetSendParameters(send_parameters_);
    for (uint32_t ssrc : ssrcs) {
      EXPECT_TRUE(
          channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(ssrc)));
    }
  
    EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
    for (uint32_t ssrc : ssrcs) {
      const auto* s = call_.GetAudioReceiveStream(ssrc);
      EXPECT_NE(nullptr, s);
      EXPECT_EQ(0u, s->GetConfig().rtp.extensions.size());
    }
  
    // Set up receive extensions.
    cricket::RtpCapabilities capabilities = engine_->GetCapabilities();
    cricket::AudioRecvParameters recv_parameters;
    recv_parameters.extensions = capabilities.header_extensions;
    channel_->SetRecvParameters(recv_parameters);
    EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
    for (uint32_t ssrc : ssrcs) {
      const auto* s = call_.GetAudioReceiveStream(ssrc);
      EXPECT_NE(nullptr, s);
      const auto& s_exts = s->GetConfig().rtp.extensions;
      EXPECT_EQ(capabilities.header_extensions.size(), s_exts.size());
      for (const auto& e_ext : capabilities.header_extensions) {
        for (const auto& s_ext : s_exts) {
          if (e_ext.id == s_ext.id) {
            EXPECT_EQ(e_ext.uri, s_ext.uri);
          }
        }
      }
    }
  
    // Disable receive extensions.
    channel_->SetRecvParameters(cricket::AudioRecvParameters());
    for (uint32_t ssrc : ssrcs) {
      const auto* s = call_.GetAudioReceiveStream(ssrc);
      EXPECT_NE(nullptr, s);
      EXPECT_EQ(0u, s->GetConfig().rtp.extensions.size());
    }
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
    // Test that packets are forwarded to the Call when configured accordingly.
    const uint32_t kAudioSsrc = 1;
    rtc::CopyOnWriteBuffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame));
    static const unsigned char kRtcp[] = {
        0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, 0x00,
        0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
        0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
    rtc::CopyOnWriteBuffer kRtcpPacket(kRtcp, sizeof(kRtcp));
  
    EXPECT_TRUE(SetupSendStream());
    cricket::WebRtcVoiceMediaChannel* media_channel =
        static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
    SetSendParameters(send_parameters_);
    EXPECT_TRUE(media_channel->AddRecvStream(
        cricket::StreamParams::CreateLegacy(kAudioSsrc)));
  
    EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
    const cricket::FakeAudioReceiveStream* s =
        call_.GetAudioReceiveStream(kAudioSsrc);
    EXPECT_EQ(0, s->received_packets());
    channel_->OnPacketReceived(kPcmuPacket, /* packet_time_us */ -1);
    EXPECT_EQ(1, s->received_packets());
    channel_->OnRtcpReceived(kRtcpPacket, /* packet_time_us */ -1);
    EXPECT_EQ(2, s->received_packets());
  }
  
  // All receive channels should be associated with the first send channel,
  // since they do not send RTCP SR.
  TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) {
    EXPECT_TRUE(SetupSendStream());
    EXPECT_TRUE(AddRecvStream(kSsrcY));
    EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
    EXPECT_TRUE(
        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcZ)));
    EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
    EXPECT_TRUE(AddRecvStream(kSsrcW));
    EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcW).rtp.local_ssrc);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_RecvCreatedFirst) {
    EXPECT_TRUE(SetupRecvStream());
    EXPECT_EQ(0xFA17FA17u, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
    EXPECT_TRUE(
        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcY)));
    EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
    EXPECT_TRUE(AddRecvStream(kSsrcZ));
    EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc);
    EXPECT_TRUE(
        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcW)));
    EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
    EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc);
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetRawAudioSink) {
    EXPECT_TRUE(SetupChannel());
    std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink());
    std::unique_ptr<FakeAudioSink> fake_sink_2(new FakeAudioSink());
  
    // Setting the sink before a recv stream exists should do nothing.
    channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_1));
    EXPECT_TRUE(AddRecvStream(kSsrcX));
    EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink());
  
    // Now try actually setting the sink.
    channel_->SetRawAudioSink(kSsrcX, std::move(fake_sink_2));
    EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink());
  
    // Now try resetting it.
    channel_->SetRawAudioSink(kSsrcX, nullptr);
    EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink());
  }
  
  TEST_F(WebRtcVoiceEngineTestFake, SetRawAudioSinkUnsignaledRecvStream) {
    EXPECT_TRUE(SetupChannel());
    std::unique_ptr<FakeAudioSink> fake_sink_1(new FakeAudioSink());
    std::unique_ptr<FakeAudioSink> fake_sink_2(new FakeAudioSink());
    std::unique_ptr<FakeAudioSink> fake_sink_3(new FakeAudioSink());
    std::unique_ptr<FakeAudioSink> fake_sink_4(new FakeAudioSink());
  
    // Should be able to set a default sink even when no stream exists.
    channel_->SetRawAudioSink(0, std::move(fake_sink_1));
  
    // Spawn an unsignaled stream by sending a packet - it should be assigned the
    // default sink.
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
    EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
  
    // Try resetting the default sink.
    channel_->SetRawAudioSink(kSsrc0, nullptr);
    EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
  
    // Try setting the default sink while the default stream exists.
    channel_->SetRawAudioSink(kSsrc0, std::move(fake_sink_2));
    EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
  
    // If we remove and add a default stream, it should get the same sink.
    EXPECT_TRUE(channel_->RemoveRecvStream(kSsrc1));
    DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
    EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
  
    // Spawn another unsignaled stream - it should be assigned the default sink
    // and the previous unsignaled stream should lose it.
    unsigned char pcmuFrame2[sizeof(kPcmuFrame)];
    memcpy(pcmuFrame2, kPcmuFrame, sizeof(kPcmuFrame));
    rtc::SetBE32(&pcmuFrame2[8], kSsrcX);
    DeliverPacket(pcmuFrame2, sizeof(pcmuFrame2));
    if (kMaxUnsignaledRecvStreams > 1) {
      EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
    }
    EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink());
  
    // Reset the default sink - the second unsignaled stream should lose it.
    channel_->SetRawAudioSink(kSsrc0, nullptr);
    if (kMaxUnsignaledRecvStreams > 1) {
      EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
    }
    EXPECT_EQ(nullptr, GetRecvStream(kSsrcX).sink());
  
    // Try setting the default sink while two streams exists.
    channel_->SetRawAudioSink(kSsrc0, std::move(fake_sink_3));
    if (kMaxUnsignaledRecvStreams > 1) {
      EXPECT_EQ(nullptr, GetRecvStream(kSsrc1).sink());
    }
    EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink());
  
    // Try setting the sink for the first unsignaled stream using its known SSRC.
    channel_->SetRawAudioSink(kSsrc1, std::move(fake_sink_4));
    if (kMaxUnsignaledRecvStreams > 1) {
      EXPECT_NE(nullptr, GetRecvStream(kSsrc1).sink());
    }
    EXPECT_NE(nullptr, GetRecvStream(kSsrcX).sink());
    if (kMaxUnsignaledRecvStreams > 1) {
      EXPECT_NE(GetRecvStream(kSsrc1).sink(), GetRecvStream(kSsrcX).sink());
    }
  }
  
  // Test that, just like the video channel, the voice channel communicates the
  // network state to the call.
  TEST_F(WebRtcVoiceEngineTestFake, OnReadyToSendSignalsNetworkState) {
    EXPECT_TRUE(SetupChannel());
  
    EXPECT_EQ(webrtc::kNetworkUp,
              call_.GetNetworkState(webrtc::MediaType::AUDIO));
    EXPECT_EQ(webrtc::kNetworkUp,
              call_.GetNetworkState(webrtc::MediaType::VIDEO));
  
    channel_->OnReadyToSend(false);
    EXPECT_EQ(webrtc::kNetworkDown,
              call_.GetNetworkState(webrtc::MediaType::AUDIO));
    EXPECT_EQ(webrtc::kNetworkUp,
              call_.GetNetworkState(webrtc::MediaType::VIDEO));
  
    channel_->OnReadyToSend(true);
    EXPECT_EQ(webrtc::kNetworkUp,
              call_.GetNetworkState(webrtc::MediaType::AUDIO));
    EXPECT_EQ(webrtc::kNetworkUp,
              call_.GetNetworkState(webrtc::MediaType::VIDEO));
  }
  
  // Test that playout is still started after changing parameters
  TEST_F(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) {
    SetupRecvStream();
    channel_->SetPlayout(true);
    EXPECT_TRUE(GetRecvStream(kSsrcX).started());
  
    // Changing RTP header extensions will recreate the AudioReceiveStream.
    cricket::AudioRecvParameters parameters;
    parameters.extensions.push_back(
        webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12));
    channel_->SetRecvParameters(parameters);
  
    EXPECT_TRUE(GetRecvStream(kSsrcX).started());
  }
  
  // Tests when GetSources is called with non-existing ssrc, it will return an
  // empty list of RtpSource without crashing.
  TEST_F(WebRtcVoiceEngineTestFake, GetSourcesWithNonExistingSsrc) {
    // Setup an recv stream with |kSsrcX|.
    SetupRecvStream();
    cricket::WebRtcVoiceMediaChannel* media_channel =
        static_cast<cricket::WebRtcVoiceMediaChannel*>(channel_);
    // Call GetSources with |kSsrcY| which doesn't exist.
    std::vector<webrtc::RtpSource> sources = media_channel->GetSources(kSsrcY);
    EXPECT_EQ(0u, sources.size());
  }
  
  // Tests that the library initializes and shuts down properly.
  TEST(WebRtcVoiceEngineTest, StartupShutdown) {
    // If the VoiceEngine wants to gather available codecs early, that's fine but
    // we never want it to create a decoder at this stage.
    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
        webrtc::CreateDefaultTaskQueueFactory();
    ::testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
    rtc::scoped_refptr<webrtc::AudioProcessing> apm =
        webrtc::AudioProcessingBuilder().Create();
    cricket::WebRtcVoiceEngine engine(
        task_queue_factory.get(), &adm,
        webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
        webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
    engine.Init();
    webrtc::RtcEventLogNullImpl event_log;
    webrtc::Call::Config call_config(&event_log);
    call_config.task_queue_factory = task_queue_factory.get();
    auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
    cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
        call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
        webrtc::CryptoOptions());
    EXPECT_TRUE(channel != nullptr);
    delete channel;
  }
  
  // Tests that reference counting on the external ADM is correct.
  TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
        webrtc::CreateDefaultTaskQueueFactory();
    ::testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
    EXPECT_CALL(adm, AddRef()).Times(3);
    EXPECT_CALL(adm, Release())
        .Times(3)
        .WillRepeatedly(Return(rtc::RefCountReleaseStatus::kDroppedLastRef));
    {
      rtc::scoped_refptr<webrtc::AudioProcessing> apm =
          webrtc::AudioProcessingBuilder().Create();
      cricket::WebRtcVoiceEngine engine(
          task_queue_factory.get(), &adm,
          webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
          webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
      engine.Init();
      webrtc::RtcEventLogNullImpl event_log;
      webrtc::Call::Config call_config(&event_log);
      call_config.task_queue_factory = task_queue_factory.get();
      auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
      cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
          call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
          webrtc::CryptoOptions());
      EXPECT_TRUE(channel != nullptr);
      delete channel;
    }
  }
  
  // Verify the payload id of common audio codecs, including CN, ISAC, and G722.
  TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
        webrtc::CreateDefaultTaskQueueFactory();
    // TODO(ossu): Why are the payload types of codecs with non-static payload
    // type assignments checked here? It shouldn't really matter.
    ::testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
    rtc::scoped_refptr<webrtc::AudioProcessing> apm =
        webrtc::AudioProcessingBuilder().Create();
    cricket::WebRtcVoiceEngine engine(
        task_queue_factory.get(), &adm,
        webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
        webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
    engine.Init();
    for (const cricket::AudioCodec& codec : engine.send_codecs()) {
      auto is_codec = [&codec](const char* name, int clockrate = 0) {
        return absl::EqualsIgnoreCase(codec.name, name) &&
               (clockrate == 0 || codec.clockrate == clockrate);
      };
      if (is_codec("CN", 16000)) {
        EXPECT_EQ(105, codec.id);
      } else if (is_codec("CN", 32000)) {
        EXPECT_EQ(106, codec.id);
      } else if (is_codec("ISAC", 16000)) {
        EXPECT_EQ(103, codec.id);
      } else if (is_codec("ISAC", 32000)) {
        EXPECT_EQ(104, codec.id);
      } else if (is_codec("G722", 8000)) {
        EXPECT_EQ(9, codec.id);
      } else if (is_codec("telephone-event", 8000)) {
        EXPECT_EQ(126, codec.id);
        // TODO(solenberg): 16k, 32k, 48k DTMF should be dynamically assigned.
        // Remove these checks once both send and receive side assigns payload
        // types dynamically.
      } else if (is_codec("telephone-event", 16000)) {
        EXPECT_EQ(113, codec.id);
      } else if (is_codec("telephone-event", 32000)) {
        EXPECT_EQ(112, codec.id);
      } else if (is_codec("telephone-event", 48000)) {
        EXPECT_EQ(110, codec.id);
      } else if (is_codec("opus")) {
        EXPECT_EQ(111, codec.id);
        ASSERT_TRUE(codec.params.find("minptime") != codec.params.end());
        EXPECT_EQ("10", codec.params.find("minptime")->second);
        ASSERT_TRUE(codec.params.find("useinbandfec") != codec.params.end());
        EXPECT_EQ("1", codec.params.find("useinbandfec")->second);
      }
    }
  }
  
  // Tests that VoE supports at least 32 channels
  TEST(WebRtcVoiceEngineTest, Has32Channels) {
    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
        webrtc::CreateDefaultTaskQueueFactory();
    ::testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
    rtc::scoped_refptr<webrtc::AudioProcessing> apm =
        webrtc::AudioProcessingBuilder().Create();
    cricket::WebRtcVoiceEngine engine(
        task_queue_factory.get(), &adm,
        webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
        webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm);
    engine.Init();
    webrtc::RtcEventLogNullImpl event_log;
    webrtc::Call::Config call_config(&event_log);
    call_config.task_queue_factory = task_queue_factory.get();
    auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
  
    cricket::VoiceMediaChannel* channels[32];
    size_t num_channels = 0;
    while (num_channels < arraysize(channels)) {
      cricket::VoiceMediaChannel* channel = engine.CreateMediaChannel(
          call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
          webrtc::CryptoOptions());
      if (!channel)
        break;
      channels[num_channels++] = channel;
    }
  
    size_t expected = arraysize(channels);
    EXPECT_EQ(expected, num_channels);
  
    while (num_channels > 0) {
      delete channels[--num_channels];
    }
  }
  
  // Test that we set our preferred codecs properly.
  TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
        webrtc::CreateDefaultTaskQueueFactory();
    // TODO(ossu): I'm not sure of the intent of this test. It's either:
    // - Check that our builtin codecs are usable by Channel.
    // - The codecs provided by the engine is usable by Channel.
    // It does not check that the codecs in the RecvParameters are actually
    // what we sent in - though it's probably reasonable to expect so, if
    // SetRecvParameters returns true.
    // I think it will become clear once audio decoder injection is completed.
    ::testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
    rtc::scoped_refptr<webrtc::AudioProcessing> apm =
        webrtc::AudioProcessingBuilder().Create();
    cricket::WebRtcVoiceEngine engine(
        task_queue_factory.get(), &adm,
        webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
        webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm);
    engine.Init();
    webrtc::RtcEventLogNullImpl event_log;
    webrtc::Call::Config call_config(&event_log);
    call_config.task_queue_factory = task_queue_factory.get();
    auto call = absl::WrapUnique(webrtc::Call::Create(call_config));
    cricket::WebRtcVoiceMediaChannel channel(&engine, cricket::MediaConfig(),
                                             cricket::AudioOptions(),
                                             webrtc::CryptoOptions(), call.get());
    cricket::AudioRecvParameters parameters;
    parameters.codecs = engine.recv_codecs();
    EXPECT_TRUE(channel.SetRecvParameters(parameters));
  }
  
  TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
    std::vector<webrtc::AudioCodecSpec> specs;
    webrtc::AudioCodecSpec spec1{{"codec1", 48000, 2, {{"param1", "value1"}}},
                                 {48000, 2, 16000, 10000, 20000}};
    spec1.info.allow_comfort_noise = false;
    spec1.info.supports_network_adaption = true;
    specs.push_back(spec1);
    webrtc::AudioCodecSpec spec2{{"codec2", 32000, 1}, {32000, 1, 32000}};
    spec2.info.allow_comfort_noise = false;
    specs.push_back(spec2);
    specs.push_back(webrtc::AudioCodecSpec{
        {"codec3", 16000, 1, {{"param1", "value1b"}, {"param2", "value2"}}},
        {16000, 1, 13300}});
    specs.push_back(
        webrtc::AudioCodecSpec{{"codec4", 8000, 1}, {8000, 1, 64000}});
    specs.push_back(
        webrtc::AudioCodecSpec{{"codec5", 8000, 2}, {8000, 1, 64000}});
  
    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
        webrtc::CreateDefaultTaskQueueFactory();
    rtc::scoped_refptr<webrtc::MockAudioEncoderFactory> unused_encoder_factory =
        webrtc::MockAudioEncoderFactory::CreateUnusedFactory();
    rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory =
        new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>;
    EXPECT_CALL(*mock_decoder_factory.get(), GetSupportedDecoders())
        .WillOnce(Return(specs));
    ::testing::NiceMock<webrtc::test::MockAudioDeviceModule> adm;
  
    rtc::scoped_refptr<webrtc::AudioProcessing> apm =
        webrtc::AudioProcessingBuilder().Create();
    cricket::WebRtcVoiceEngine engine(task_queue_factory.get(), &adm,
                                      unused_encoder_factory,
                                      mock_decoder_factory, nullptr, apm);
    engine.Init();
    auto codecs = engine.recv_codecs();
    EXPECT_EQ(11u, codecs.size());
  
    // Rather than just ASSERTing that there are enough codecs, ensure that we can
    // check the actual values safely, to provide better test results.
    auto get_codec = [&codecs](size_t index) -> const cricket::AudioCodec& {
      static const cricket::AudioCodec missing_codec(0, "<missing>", 0, 0, 0);
      if (codecs.size() > index)
        return codecs[index];
      return missing_codec;
    };
  
    // Ensure the general codecs are generated first and in order.
    for (size_t i = 0; i != specs.size(); ++i) {
      EXPECT_EQ(specs[i].format.name, get_codec(i).name);
      EXPECT_EQ(specs[i].format.clockrate_hz, get_codec(i).clockrate);
      EXPECT_EQ(specs[i].format.num_channels, get_codec(i).channels);
      EXPECT_EQ(specs[i].format.parameters, get_codec(i).params);
    }
  
    // Find the index of a codec, or -1 if not found, so that we can easily check
    // supplementary codecs are ordered after the general codecs.
    auto find_codec = [&codecs](const webrtc::SdpAudioFormat& format) -> int {
      for (size_t i = 0; i != codecs.size(); ++i) {
        const cricket::AudioCodec& codec = codecs[i];
        if (absl::EqualsIgnoreCase(codec.name, format.name) &&
            codec.clockrate == format.clockrate_hz &&
            codec.channels == format.num_channels) {
          return rtc::checked_cast<int>(i);
        }
      }
      return -1;
    };
  
    // Ensure all supplementary codecs are generated last. Their internal ordering
    // is not important.
    // Without this cast, the comparison turned unsigned and, thus, failed for -1.
    const int num_specs = static_cast<int>(specs.size());
    EXPECT_GE(find_codec({"cn", 8000, 1}), num_specs);
    EXPECT_GE(find_codec({"cn", 16000, 1}), num_specs);
    EXPECT_EQ(find_codec({"cn", 32000, 1}), -1);
    EXPECT_GE(find_codec({"telephone-event", 8000, 1}), num_specs);
    EXPECT_GE(find_codec({"telephone-event", 16000, 1}), num_specs);
    EXPECT_GE(find_codec({"telephone-event", 32000, 1}), num_specs);
    EXPECT_GE(find_codec({"telephone-event", 48000, 1}), num_specs);
  }
  