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ABSL_FLAG(int, width, 1280, "Video width.");
ABSL_FLAG(int, height, 720, "Video height.");
ABSL_FLAG(int, fps, 30, "Frames per second.");
ABSL_FLAG(int, capture_device_index, 0, "Capture device to select");
ABSL_FLAG(int, min_bitrate, 1000, "Call and stream min bitrate in kbps.");
ABSL_FLAG(int, start_bitrate, 1000, "Call start bitrate in kbps.");
ABSL_FLAG(int, target_bitrate, 1200, "Stream target bitrate in kbps.");
ABSL_FLAG(int, max_bitrate, 1500, "Call and stream max bitrate in kbps.");

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1 main
2 webrtc::RunLoopbackTest
3 void Loopback
4 VideoQualityTest::RunWithRenderers

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void VideoQualityTest::RunWithRenderers(const Params& params) {
  RTC_LOG(INFO) << __FUNCTION__;
  num_video_streams_ = params.call.dual_video ? 2 : 1;
  std::unique_ptr<test::LayerFilteringTransport> send_transport;
  std::unique_ptr<test::DirectTransport> recv_transport;
  std::unique_ptr<test::VideoRenderer> local_preview;
  std::vector<std::unique_ptr<test::VideoRenderer>> loopback_renderers;

  loopback_renderers.emplace_back(
      test::VideoRenderer::Create("Loopback Video1", 640, 480));
  local_preview.reset(test::VideoRenderer::Create("Local Preview", 640, 480));

  if (!params.logging.rtc_event_log_name.empty()) {
    send_event_log_ = rtc_event_log_factory_.CreateRtcEventLog(
        RtcEventLog::EncodingType::Legacy);
    recv_event_log_ = rtc_event_log_factory_.CreateRtcEventLog(
        RtcEventLog::EncodingType::Legacy);
    std::unique_ptr<RtcEventLogOutputFile> send_output(
        std::make_unique<RtcEventLogOutputFile>(
            params.logging.rtc_event_log_name + "_send",
            RtcEventLog::kUnlimitedOutput));
    std::unique_ptr<RtcEventLogOutputFile> recv_output(
        std::make_unique<RtcEventLogOutputFile>(
            params.logging.rtc_event_log_name + "_recv",
            RtcEventLog::kUnlimitedOutput));
    bool event_log_started =
        send_event_log_->StartLogging(std::move(send_output),
                                      /*output_period_ms=*/5000) &&
        recv_event_log_->StartLogging(std::move(recv_output),
                                      /*output_period_ms=*/5000);
    RTC_DCHECK(event_log_started);
  } else {
    send_event_log_ = std::make_unique<RtcEventLogNull>();
    recv_event_log_ = std::make_unique<RtcEventLogNull>();
  }

  SendTask(RTC_FROM_HERE, task_queue(), [&]() {
    params_ = params;
    CheckParamsAndInjectionComponents();

    // TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
    // match the full stack tests.
    Call::Config send_call_config(send_event_log_.get());
    send_call_config.bitrate_config = params_.call.call_bitrate_config;
    Call::Config recv_call_config(recv_event_log_.get());

    if (params_.audio.enabled)
      InitializeAudioDevice(&send_call_config, &recv_call_config,
                            params_.audio.use_real_adm);

    CreateCalls(send_call_config, recv_call_config);

    // TODO(minyue): consider if this is a good transport even for audio only
    // calls.
    send_transport = CreateSendTransport();

    recv_transport = CreateReceiveTransport();

    // TODO(ivica): Use two calls to be able to merge with RunWithAnalyzer or at
    // least share as much code as possible. That way this test would also match
    // the full stack tests better.
    send_transport->SetReceiver(receiver_call_->Receiver());
    recv_transport->SetReceiver(sender_call_->Receiver());

    if (params_.video[0].enabled) {
      // Create video renderers.
      SetupVideo(send_transport.get(), recv_transport.get());
      size_t num_streams_processed = 0;
      for (size_t video_idx = 0; video_idx < num_video_streams_; ++video_idx) {
        const size_t selected_stream_id = params_.ss[video_idx].selected_stream;
        const size_t num_streams = params_.ss[video_idx].streams.size();
        if (selected_stream_id == num_streams) {
          for (size_t stream_id = 0; stream_id < num_streams; ++stream_id) {
            rtc::StringBuilder oss;
            oss << "Loopback Video #" << video_idx << " - Stream #"
                << static_cast<int>(stream_id);
            /* loopback_renderers.emplace_back(test::VideoRenderer::Create(
                oss.str().c_str(),
                params_.ss[video_idx].streams[stream_id].width,
                params_.ss[video_idx].streams[stream_id].height));*/
          
            video_receive_configs_[stream_id + num_streams_processed].renderer =
                loopback_renderers.back().get();
            if (params_.audio.enabled && params_.audio.sync_video)
              video_receive_configs_[stream_id + num_streams_processed]
                  .sync_group = kSyncGroup;
          }
        } else {
          rtc::StringBuilder oss;
          oss << "Loopback Video #" << video_idx;
          /* loopback_renderers.emplace_back(test::VideoRenderer::Create(
              oss.str().c_str(),
              params_.ss[video_idx].streams[selected_stream_id].width,
              params_.ss[video_idx].streams[selected_stream_id].height));*/

          video_receive_configs_[selected_stream_id + num_streams_processed]
              .renderer = loopback_renderers.back().get();
          if (params_.audio.enabled && params_.audio.sync_video)
            video_receive_configs_[num_streams_processed + selected_stream_id]
                .sync_group = kSyncGroup;
        }
        num_streams_processed += num_streams;
      }
      CreateFlexfecStreams();
      CreateVideoStreams();

      CreateCapturers();
      if (params_.video[0].enabled) {
        // Create local preview
       /* local_preview.reset(test::VideoRenderer::Create(
            "Local Preview", params_.video[0].width, params_.video[0].height));*/

        video_sources_[0]->AddOrUpdateSink(local_preview.get(),
                                           rtc::VideoSinkWants());
      }
      ConnectVideoSourcesToStreams();
    }

    if (params_.audio.enabled) {
      SetupAudio(send_transport.get());
    }

    Start();
  });

   MSG msg;
  while (GetMessage(&msg, NULL, 0, 0) > 0) {
    TranslateMessage(&msg);
    DispatchMessage(&msg);
  }

  PressEnterToContinue(task_queue());

  SendTask(RTC_FROM_HERE, task_queue(), [&]() {
    Stop();
    DestroyStreams();

    send_transport.reset();
    recv_transport.reset();

    local_preview.reset();
    loopback_renderers.clear();

    DestroyCalls();
  });
}

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  std::unique_ptr<test::VideoRenderer> local_preview;
  std::vector<std::unique_ptr<test::VideoRenderer>> loopback_renderers;
  loopback_renderers.emplace_back(
      test::VideoRenderer::Create("Loopback Video1", 640, 480));
  local_preview.reset(test::VideoRenderer::Create("Local Preview", 640, 480));

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    Call::Config send_call_config(send_event_log_.get());
    send_call_config.bitrate_config = params_.call.call_bitrate_config;
    Call::Config recv_call_config(recv_event_log_.get());

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// TODO(srte): BitrateConstraints and BitrateSettings should be merged.
// Both represent the same kind data, but are using different default
// initializer and representation of unset values.
struct BitrateConstraints {
 int min_bitrate_bps = 0;
 int start_bitrate_bps = kDefaultStartBitrateBps;
 int max_bitrate_bps = -1;

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void VideoQualityTest::InitializeAudioDevice(Call::Config* send_call_config,
                                             Call::Config* recv_call_config,
                                             bool use_real_adm) {
  rtc::scoped_refptr<AudioDeviceModule> audio_device;
  if (use_real_adm) {
    // Run test with real ADM (using default audio devices) if user has
    // explicitly set the --audio and --use_real_adm command-line flags.
    audio_device = CreateAudioDevice();
  } else {
    // By default, create a test ADM which fakes audio.
    audio_device = TestAudioDeviceModule::Create(
        task_queue_factory_.get(),
        TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, 48000),
        TestAudioDeviceModule::CreateDiscardRenderer(48000), 1.f);
  }
  RTC_CHECK(audio_device);

  AudioState::Config audio_state_config;
  audio_state_config.audio_mixer = AudioMixerImpl::Create();
  audio_state_config.audio_processing = AudioProcessingBuilder().Create();
  audio_state_config.audio_device_module = audio_device;
  send_call_config->audio_state = AudioState::Create(audio_state_config);
  recv_call_config->audio_state = AudioState::Create(audio_state_config);
  if (use_real_adm) {
    // The real ADM requires extra initialization: setting default devices,
    // setting up number of channels etc. Helper class also calls
    // AudioDeviceModule::Init().
    webrtc::adm_helpers::Init(audio_device.get());
  } else {
    audio_device->Init();
  }
  // Always initialize the ADM before injecting a valid audio transport.
  RTC_CHECK(audio_device->RegisterAudioCallback(
                send_call_config->audio_state->audio_transport()) == 0);
}

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  // Device enumeration
  virtual int16_t PlayoutDevices() = 0;
  virtual int16_t RecordingDevices() = 0;
  virtual int32_t PlayoutDeviceName(uint16_t index,
                                    char name[kAdmMaxDeviceNameSize],
                                    char guid[kAdmMaxGuidSize]) = 0;
  virtual int32_t RecordingDeviceName(uint16_t index,
                                      char name[kAdmMaxDeviceNameSize],
                                      char guid[kAdmMaxGuidSize]) = 0;

  // Device selection
  virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
  virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
  virtual int32_t SetRecordingDevice(uint16_t index) = 0;
  virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;

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    AudioState::AudioState(const AudioState::Config& config)
    : config_(config),
      audio_transport_(config_.audio_mixer,
                       config_.audio_processing.get(),
                       config_.async_audio_processing_factory.get()) {
  process_thread_checker_.Detach();
  RTC_DCHECK(config_.audio_mixer);
  RTC_DCHECK(config_.audio_device_module);
}

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class AudioTransport {
 public:
 //adm mic 采集pcm 通过调用这接口进入apm
  virtual int32_t RecordedDataIsAvailable(const void* audioSamples,
                                          const size_t nSamples,
                                          const size_t nBytesPerSample,
                                          const size_t nChannels,
                                          const uint32_t samplesPerSec,
                                          const uint32_t totalDelayMS,
                                          const int32_t clockDrift,
                                          const uint32_t currentMicLevel,
                                          const bool keyPressed,
                                          uint32_t& newMicLevel) = 0;  // NOLINT
  //adm 播放时，从这接口获取pcm数据
  // Implementation has to setup safe values for all specified out parameters.
  virtual int32_t NeedMorePlayData(const size_t nSamples,
                                   const size_t nBytesPerSample,
                                   const size_t nChannels,
                                   const uint32_t samplesPerSec,
                                   void* audioSamples,
                                   size_t& nSamplesOut,  // NOLINT
                                   int64_t* elapsed_time_ms,
                                   int64_t* ntp_time_ms) = 0;  // NOLINT

  // Method to pull mixed render audio data from all active VoE channels.
  // The data will not be passed as reference for audio processing internally.
  virtual void PullRenderData(int bits_per_sample,
                              int sample_rate,
                              size_t number_of_channels,
                              size_t number_of_frames,
                              void* audio_data,
                              int64_t* elapsed_time_ms,
                              int64_t* ntp_time_ms) = 0;

 protected:
  virtual ~AudioTransport() {}
};

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void CallTest::CreateSenderCall(const Call::Config& config) {
  auto sender_config = config;
  sender_config.task_queue_factory = task_queue_factory_.get();
  sender_config.network_state_predictor_factory =
      network_state_predictor_factory_.get();
  sender_config.network_controller_factory = network_controller_factory_.get();
  sender_config.trials = &field_trials_;
  sender_call_.reset(Call::Create(sender_config));
}

void CallTest::CreateReceiverCall(const Call::Config& config) {
  auto receiver_config = config;
  receiver_config.task_queue_factory = task_queue_factory_.get();
  receiver_config.trials = &field_trials_;
  receiver_call_.reset(Call::Create(receiver_config));
}

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send_transport = CreateSendTransport();
recv_transport = CreateReceiveTransport();

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class DirectTransport : public Transport {
 public:
 ...
  // TODO(holmer): Look into moving this to the constructor.
  virtual void SetReceiver(PacketReceiver* receiver);

  bool SendRtp(const uint8_t* data,
               size_t length,
               const PacketOptions& options) override;
  bool SendRtcp(const uint8_t* data, size_t length) override;
...
};

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//网络相关，payloadtype，nack
std::vector<VideoSendStream::Config> video_send_configs_;
//编码器相关
std::vector<VideoEncoderConfig> video_encoder_configs_;
std::vector<VideoReceiveStream::Config> video_receive_configs_;

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void VideoQualityTest::SetupVideo(Transport* send_transport,
                                  Transport* recv_transport) {
  ......
  video_receive_configs_.clear();
  video_send_configs_.clear();
  video_encoder_configs_.clear();
  // 一个视频是可以发多个流，类似svc的实现，但这里不是svc分层，是通过编码了多个分辨率的流，平时我们只用一个
  for (size_t video_idx = 0; video_idx < num_video_streams_; ++video_idx) {
    //设置视频流的网络传输transport
    video_send_configs_.push_back(VideoSendStream::Config(send_transport));
    video_encoder_configs_.push_back(VideoEncoderConfig());
    num_video_substreams = params_.ss[video_idx].streams.size();
    RTC_CHECK_GT(num_video_substreams, 0);
    for (size_t i = 0; i < num_video_substreams; ++i) {
      //rtp ssrc设置，用于标识rtp流，详情看协议
      video_send_configs_[video_idx].rtp.ssrcs.push_back(
          kVideoSendSsrcs[total_streams_used + i]);
    }
    ......
    //设置rtp中的pt，用于接收端接收到rtp后，rtp的组帧方式和解码器的选取
    if (params_.video[video_idx].codec == "H264") {
      payload_type = kPayloadTypeH264;
    }
    ...... 
    //设置编码器factory
     video_send_configs_[video_idx].encoder_settings.encoder_factory =
        (video_idx == 0) ? &video_encoder_factory_with_analyzer_
                         : &video_encoder_factory_;
    //码率控制      factory              
    video_send_configs_[video_idx].encoder_settings.bitrate_allocator_factory =
        video_bitrate_allocator_factory_.get();
   ......
   //编码器名字
    video_send_configs_[video_idx].rtp.payload_name =
        params_.video[video_idx].codec;
    //rtp payload type
    video_send_configs_[video_idx].rtp.payload_type = payload_type;
    //nack 历史队列保存时间，不为0 时代表nack开启
    video_send_configs_[video_idx].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
    video_send_configs_[video_idx].rtp.rtx.payload_type = kSendRtxPayloadType;
     // 发送端代码估计，transport-cc需要开启这个,索引对应必须和接收端一致，否则可能引起崩溃
	 if (params_.call.send_side_bwe) {
      video_send_configs_[video_idx].rtp.extensions.emplace_back(
          RtpExtension::kTransportSequenceNumberUri,
          kTransportSequenceNumberExtensionId);
    } else {
      //只接收也能计算rtt，接收端估计
      video_send_configs_[video_idx].rtp.extensions.emplace_back(
          RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId);
    }
    ......
    //编码参数设置
    video_encoder_configs_[video_idx].video_format.name =
        params_.video[video_idx].codec;
    video_encoder_configs_[video_idx].max_bitrate_bps = 0;
    for (size_t i = 0; i < params_.ss[video_idx].streams.size(); ++i) {
      video_encoder_configs_[video_idx].max_bitrate_bps +=
          params_.ss[video_idx].streams[i].max_bitrate_bps;
    }  
    ......  
    video_encoder_configs_[video_idx].video_stream_factory =
        rtc::make_ref_counted<cricket::EncoderStreamFactory>(
            params_.video[video_idx].codec,
            params_.ss[video_idx].streams[0].max_qp,
            params_.screenshare[video_idx].enabled, true);

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void CallTest::CreateVideoStreams() {
  RTC_DCHECK(video_receive_streams_.empty());
  CreateVideoSendStreams();
  for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
    video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
        video_receive_configs_[i].Copy()));
  }
}

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void CallTest::CreateVideoSendStreams() {
......
  for (size_t i : streams_creation_order) {
    if (fec_controller_factory_.get()) {
      video_send_streams_[i] = sender_call_->CreateVideoSendStream(
          video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy(),
          fec_controller_factory_->CreateFecController());
    } else {
      video_send_streams_[i] = sender_call_->CreateVideoSendStream(
          video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy());
    }
  }
  ......
}

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void CallTest::ConnectVideoSourcesToStreams() {
  for (size_t i = 0; i < video_sources_.size(); ++i)
    video_send_streams_[i]->SetSource(video_sources_[i].get(),
                                      degradation_preference_);
}

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void VideoQualityTest::SetupAudio(Transport* transport) {
  AudioSendStream::Config audio_send_config(transport);
  audio_send_config.rtp.ssrc = kAudioSendSsrc;

  // Add extension to enable audio send side BWE, and allow audio bit rate
  // adaptation.
  audio_send_config.rtp.extensions.clear();
  audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
      kAudioSendPayloadType,
      {"OPUS",
       48000,
       2,
       {{"usedtx", (params_.audio.dtx ? "1" : "0")}, {"stereo", "1"}}});

  if (params_.call.send_side_bwe) {
    audio_send_config.rtp.extensions.push_back(
        webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri,
                             kTransportSequenceNumberExtensionId));
    audio_send_config.min_bitrate_bps = kOpusMinBitrateBps;
    audio_send_config.max_bitrate_bps = kOpusBitrateFbBps;
    audio_send_config.send_codec_spec->transport_cc_enabled = true;
    // Only allow ANA when send-side BWE is enabled.
    audio_send_config.audio_network_adaptor_config = params_.audio.ana_config;
  }
  audio_send_config.encoder_factory = audio_encoder_factory_;
  SetAudioConfig(audio_send_config);

//音视频同步设置，只要音视频设置的字符串一样 就会配对
  std::string sync_group;
  if (params_.video[0].enabled && params_.audio.sync_video)
    sync_group = kSyncGroup;

  CreateMatchingAudioConfigs(transport, sync_group);
  CreateAudioStreams();
}

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AudioReceiveStream::Config CallTest::CreateMatchingAudioConfig(
    const AudioSendStream::Config& send_config,
    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
    Transport* transport,
    std::string sync_group) {
  AudioReceiveStream::Config audio_config;
  audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
  audio_config.rtcp_send_transport = transport;
  //接收的流的ssrc，只会接收这一个流，是对应的
  audio_config.rtp.remote_ssrc = send_config.rtp.ssrc;
  audio_config.rtp.transport_cc =
      send_config.send_codec_spec
          ? send_config.send_codec_spec->transport_cc_enabled
          : false;
  audio_config.rtp.extensions = send_config.rtp.extensions;
  audio_config.decoder_factory = audio_decoder_factory;
  //这个需要和发送的对应，根据paylaod和解码器的对应关系
  audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}};
  //音视频同步配对
  audio_config.sync_group = sync_group;
  return audio_config;
}

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void CallTest::CreateAudioStreams() {
  RTC_DCHECK(audio_send_stream_ == nullptr);
  RTC_DCHECK(audio_receive_streams_.empty());
  audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
  for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
    audio_receive_streams_.push_back(
        receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
  }
}

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void CallTest::Start() {
  StartVideoStreams();
  if (audio_send_stream_) {
    audio_send_stream_->Start();
  }
  for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
    audio_recv_stream->Start();
}

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void CallTest::StartVideoStreams() {
  for (VideoSendStream* video_send_stream : video_send_streams_)
    video_send_stream->Start();
  for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
    video_recv_stream->Start();
}

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  std::vector<std::unique_ptr<rtc::VideoSourceInterface<VideoFrame>>>
      thumbnail_capturers_;
  Clock* const clock_;
  const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
  RtcEventLogFactory rtc_event_log_factory_;

  test::FunctionVideoDecoderFactory video_decoder_factory_;
  std::unique_ptr<VideoDecoderFactory> decoder_factory_;
  test::FunctionVideoEncoderFactory video_encoder_factory_;
  test::FunctionVideoEncoderFactory video_encoder_factory_with_analyzer_;
  std::unique_ptr<VideoBitrateAllocatorFactory>
      video_bitrate_allocator_factory_;
  std::unique_ptr<VideoEncoderFactory> encoder_factory_;
  std::vector<VideoSendStream::Config> thumbnail_send_configs_;
  std::vector<VideoEncoderConfig> thumbnail_encoder_configs_;
  std::vector<VideoSendStream*> thumbnail_send_streams_;
  std::vector<VideoReceiveStream::Config> thumbnail_receive_configs_;
  std::vector<VideoReceiveStream*> thumbnail_receive_streams_;

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class VideoSourceInterface {
public:
 virtual ~VideoSourceInterface() = default;

 virtual void AddOrUpdateSink(VideoSinkInterface<VideoFrameT>* sink,
                              const VideoSinkWants& wants) = 0;
 // RemoveSink must guarantee that at the time the method returns,
 // there is no current and no future calls to VideoSinkInterface::OnFrame.
 virtual void RemoveSink(VideoSinkInterface<VideoFrameT>* sink) = 0;
};

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class VideoSinkInterface {
 public:
  virtual ~VideoSinkInterface() = default;
  virtual void OnFrame(const VideoFrameT& frame) = 0;
  // Should be called by the source when it discards the frame due to rate
  // limiting.
  virtual void OnDiscardedFrame() {}
};

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  test::FunctionVideoDecoderFactory video_decoder_factory_;
  std::unique_ptr<VideoDecoderFactory> decoder_factory_;

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class RTC_EXPORT VideoDecoderFactory {
 public:
  struct CodecSupport {
    bool is_supported = false;
    bool is_power_efficient = false;
  };
  virtual std::vector<SdpVideoFormat> GetSupportedFormats() const = 0;
  virtual CodecSupport QueryCodecSupport(const SdpVideoFormat& format,
                                         bool reference_scaling) const {
    CodecSupport codec_support;
    codec_support.is_supported =
        !reference_scaling && format.IsCodecInList(GetSupportedFormats());
    return codec_support;
  }

  virtual CodecSupport QueryCodecSupport(
      const SdpVideoFormat& format,
      absl::optional<std::string> scalability_mode) const {
    CodecSupport codec_support;
    if (!scalability_mode) {
      codec_support.is_supported = format.IsCodecInList(GetSupportedFormats());
    }
    return codec_support;
  }
  // Creates a VideoDecoder for the specified format.
  virtual std::unique_ptr<VideoDecoder> CreateVideoDecoder(
      const SdpVideoFormat& format) = 0;
  virtual ~VideoDecoderFactory() {}
};

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virtual std::unique_ptr<VideoDecoder> CreateVideoDecoder(
      const SdpVideoFormat& format) = 0;

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VideoQualityTest::VideoQualityTest(
    std::unique_ptr<InjectionComponents> injection_components)
    ......
{
  if (injection_components_ == nullptr) {
    injection_components_ = std::make_unique<InjectionComponents>();
  }
  if (injection_components_->video_decoder_factory != nullptr) {
    decoder_factory_ = std::move(injection_components_->video_decoder_factory);
  } else {
    decoder_factory_ = std::make_unique<InternalDecoderFactory>();
    }
  ......

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std::unique_ptr<VideoDecoder> VideoQualityTest::CreateVideoDecoder(
    const SdpVideoFormat& format) {
  std::unique_ptr<VideoDecoder> decoder;
  if (format.name == "multiplex") {
    decoder = std::make_unique<MultiplexDecoderAdapter>(
        decoder_factory_.get(), SdpVideoFormat(cricket::kVp9CodecName));
  } else if (format.name == "FakeCodec") {
    decoder = webrtc::FakeVideoDecoderFactory::CreateVideoDecoder();
  } else {
    decoder = decoder_factory_->CreateVideoDecoder(format);
  }
  ......

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std::unique_ptr<VideoDecoder> InternalDecoderFactory::CreateVideoDecoder(
    const SdpVideoFormat& format) {
  if (!format.IsCodecInList(GetSupportedFormats())) {
    return nullptr;
  }
  if (absl::EqualsIgnoreCase(format.name, cricket::kVp8CodecName))
    return VP8Decoder::Create();
  if (absl::EqualsIgnoreCase(format.name, cricket::kVp9CodecName))
    return VP9Decoder::Create();
  if (absl::EqualsIgnoreCase(format.name, cricket::kH264CodecName))
    return H264Decoder::Create();
  if (kIsLibaomAv1DecoderSupported &&
      absl::EqualsIgnoreCase(format.name, cricket::kAv1CodecName))
    return CreateLibaomAv1Decoder();

  RTC_NOTREACHED();
  return nullptr;
}

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if (injection_components_->video_encoder_factory != nullptr) {
    encoder_factory_ = std::move(injection_components_->video_encoder_factory);
  } else {
    encoder_factory_ = std::make_unique<InternalEncoderFactory>();
  }

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std::unique_ptr<VideoEncoder> VideoQualityTest::CreateVideoEncoder(
    const SdpVideoFormat& format,
    VideoAnalyzer* analyzer) {
  std::unique_ptr<VideoEncoder> encoder;
  if (format.name == "VP8") {
    encoder =
        std::make_unique<EncoderSimulcastProxy>(encoder_factory_.get(), format);
  } else if (format.name == "multiplex") {
    encoder = std::make_unique<MultiplexEncoderAdapter>(
        encoder_factory_.get(), SdpVideoFormat(cricket::kVp9CodecName));
  } else if (format.name == "FakeCodec") {
    encoder = webrtc::FakeVideoEncoderFactory::CreateVideoEncoder();
  } else {
    encoder = encoder_factory_->CreateVideoEncoder(format);
  }

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std::unique_ptr<VideoEncoder> InternalEncoderFactory::CreateVideoEncoder(
    const SdpVideoFormat& format) {
  if (absl::EqualsIgnoreCase(format.name, cricket::kH264CodecName))
    return H264Encoder::Create(cricket::VideoCodec(format));
  if (absl::EqualsIgnoreCase(format.name, cricket::kVp8CodecName))
    return VP8Encoder::Create();
  if (absl::EqualsIgnoreCase(format.name, cricket::kVp9CodecName))
    return VP9Encoder::Create(cricket::VideoCodec(format));
  if (kIsLibaomAv1EncoderSupported &&
      absl::EqualsIgnoreCase(format.name, cricket::kAv1CodecName))
    return CreateLibaomAv1Encoder();
  RTC_LOG(LS_ERROR) << "Trying to created encoder of unsupported format "
                    << format.name;
  return nullptr;
}

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class FunctionVideoEncoderFactory final : public VideoEncoderFactory {
 public:
  explicit FunctionVideoEncoderFactory(
      std::function<std::unique_ptr<VideoEncoder>()> create)
      : create_([create = std::move(create)](const SdpVideoFormat&) {
          return create();
        }) {}
  explicit FunctionVideoEncoderFactory(
      std::function<std::unique_ptr<VideoEncoder>(const SdpVideoFormat&)>
          create)
      : create_(std::move(create)) {}

  // Unused by tests.
  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
    RTC_NOTREACHED();
    return {};
  }

  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
      const SdpVideoFormat& format) override {
    return create_(format);
  }

 private:
  const std::function<std::unique_ptr<VideoEncoder>(const SdpVideoFormat&)>
      create_;
};

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 video_decoder_factory_([this](const SdpVideoFormat& format) {
        return this->CreateVideoDecoder(format);
      }),
 video_encoder_factory_([this](const SdpVideoFormat& format) {
        return this->CreateVideoEncoder(format, nullptr);
      }),