Evaluate webrtc GCC congestion control on ns3

Soonyang Zhang

已于 2022-08-26 10:57:18 修改

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分类专栏：拥塞控制仿真 webrtc 文章标签： c++

于 2020-07-09 22:04:22 首次发布

本文链接：https://blog.csdn.net/u010643777/article/details/107237315

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Congestion control algorithms are rarely parameter-free: parameters controlling the speed the cwnd increment or the threshold to indicating link in congestion. These parameters are often considered nuisances.

Introduction

我主要关注的是webrtc中的拥塞控制算法(Google Congestion Control)。本文在ns3平台上对GCC算法进行仿真测试。
我分析的webrtc代码版本是m84。
这个博客好像对一些同学有用，但是新手会问我怎么下载webrtc代码。需要淘宝上租用vps服务器。在远端主机中执行以下命令：

mkdir webrtc-checkout
cd webrtc-checkout
fetch --nohooks webrtc
gclient sync
cd src
git checkout -b m84 refs/remotes/branch-heads/4147
gclient sync

给yuv库中的basic_types.h(third_party/libyuv/include/libyuv)增加一个编译标志。

#define LIBYUV_LEGACY_TYPES

需要安装clang++，参考博客: ubuntu 18.04 install clang 11。编译：

gn gen out/m84 --args='is_debug=false is_component_build=false is_clang=true rtc_include_tests=false rtc_use_h264=true rtc_enable_protobuf=false use_rtti=true use_custom_libcxx=false treat_warnings_as_errors=false use_ozone=true'

生成libwebrtc:

ninja -C out/m84

webrtc代码库中实现了一个简单的离散事件仿真器(webrtc/test/scenario)，但是好像只能模拟两个点(EmulatedNetworkNode)。

//third_party/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
CallClient* CreateVideoSendingClient(
    Scenario* s,
    CallClientConfig config,
    std::vector<EmulatedNetworkNode*> send_link,
    std::vector<EmulatedNetworkNode*> return_link) {
  auto* client = s->CreateClient("send", std::move(config));
  auto* route = s->CreateRoutes(client, send_link,
                                s->CreateClient("return", CallClientConfig()),
                                return_link);
  s->CreateVideoStream(route->forward(), VideoStreamConfig());
  return client;
}

需要从上面的函数看起。
本文按照视频数据的产生，编码，传输的顺序进行分析。

Data Generator

一般来说，仿真是没有必要从摄像头读取数据的。最简单的办法就是根据视频长宽，帧率，使用一个FakeFrameGenerator生成模拟数据。编码器就需要使用FakeEncoder，直接按照拥塞控制探测到的带宽，产生数据流。也可以从文件中读取YUV帧，采用真正的编码器进行处理。

//third_party/webrtc/test/scenario/scenario.cc
VideoStreamPair* Scenario::CreateVideoStream(
    std::pair<CallClient*, CallClient*> clients,
    std::function<void(VideoStreamConfig*)> config_modifier) {
  VideoStreamConfig config;
  config_modifier(&config);
  return CreateVideoStream(clients, config);
}

VideoStreamPair* Scenario::CreateVideoStream(
    std::pair<CallClient*, CallClient*> clients,
    VideoStreamConfig config) {
  video_streams_.emplace_back(
      new VideoStreamPair(clients.first, clients.second, config));
  return video_streams_.back().get();
}
//third_party/webrtc/test/scenario/video_stream.cc
VideoStreamPair::VideoStreamPair(CallClient* sender,
                                 CallClient* receiver,
                                 VideoStreamConfig config)
    : config_(config),
      matcher_(config.hooks.frame_pair_handlers),
      send_stream_(sender, config, sender->transport_.get(), &matcher_),
      receive_stream_(receiver,
                      config,
                      &send_stream_,
                      /*chosen_stream=*/0,
                      receiver->transport_.get(),
                      &matcher_) {}
//third_party/webrtc/test/scenario/video_stream.cc
SendVideoStream::SendVideoStream(CallClient* sender,
                                 VideoStreamConfig config,
                                 Transport* send_transport,
                                 VideoFrameMatcher* matcher){
   //FrameGeneratorCapturer  ?
  video_capturer_ = std::make_unique<FrameGeneratorCapturer>(
      sender_->clock_, CreateFrameGenerator(sender_->clock_, config.source),
      config.source.framerate,
      *sender->time_controller_->GetTaskQueueFactory());
  video_capturer_->Init();

  using Encoder = VideoStreamConfig::Encoder;
  using Codec = VideoStreamConfig::Encoder::Codec;
  switch (config.encoder.implementation) {
  //配置编码器工厂
    case Encoder::Implementation::kFake:
      encoder_factory_ =
          std::make_unique<FunctionVideoEncoderFactory>([this]() {
            MutexLock lock(&mutex_);
            std::unique_ptr<FakeEncoder> encoder;
            if (config_.encoder.codec == Codec::kVideoCodecVP8) {
              encoder = std::make_unique<test::FakeVp8Encoder>(sender_->clock_);
            } else if (config_.encoder.codec == Codec::kVideoCodecGeneric) {
              encoder = std::make_unique<test::FakeEncoder>(sender_->clock_);
            } else {
              RTC_NOTREACHED();
            }
            fake_encoders_.push_back(encoder.get());
            if (config_.encoder.fake.max_rate.IsFinite())
              encoder->SetMaxBitrate(config_.encoder.fake.max_rate.kbps());
            return encoder;
          });
      break;
    case VideoStreamConfig::Encoder::Implementation::kSoftware:
      encoder_factory_.reset(new InternalEncoderFactory());
      break;
    case VideoStreamConfig::Encoder::Implementation::kHardware:
      encoder_factory_ = CreateHardwareEncoderFactory();
      break;
  }
 // 在第二部分  编码  分析
}

配置视频产生器：

//third_party/webrtc/test/scenario/video_stream.cc
std::unique_ptr<FrameGeneratorInterface> CreateFrameGenerator(
    Clock* clock,
    VideoStreamConfig::Source source) {
  using Capture = VideoStreamConfig::Source::Capture;
  switch (source.capture) {
    case Capture::kGenerator:
      return CreateSquareFrameGenerator(
          source.generator.width, source.generator.height,
          source.generator.pixel_format, /*num_squares*/ absl::nullopt);
    case Capture::kVideoFile:
      RTC_CHECK(source.video_file.width && source.video_file.height);
      return CreateFromYuvFileFrameGenerator(
          {TransformFilePath(source.video_file.name)}, source.video_file.width,
          source.video_file.height, /*frame_repeat_count*/ 1);
    case Capture::kGenerateSlides:
      return CreateSlideFrameGenerator(
          source.slides.generator.width, source.slides.generator.height,
          source.slides.change_interval.seconds<double>() * source.framerate);
    case Capture::kImageSlides:
      return CreateImageSlideGenerator(clock, source.slides, source.framerate);
  }
}

一个假的视频生成器。

//third_party/webrtc/api/test/create_frame_generator.cc
std::unique_ptr<FrameGeneratorInterface> CreateSquareFrameGenerator(
    int width,
    int height,
    absl::optional<FrameGeneratorInterface::OutputType> type,
    absl::optional<int> num_squares) {
  return std::make_unique<SquareGenerator>(
      width, height, type.value_or(FrameGeneratorInterface::OutputType::kI420),
      num_squares.value_or(10));
}

视频捕获器按照帧率，不断生成模拟器产生的视频数据。

void FrameGeneratorCapturer::Start() {
  {
    rtc::CritScope cs(&lock_);
    sending_ = true;
  }
  if (!frame_task_.Running()) {
    frame_task_ = RepeatingTaskHandle::Start(task_queue_.Get(), [this] {
      InsertFrame();
      return TimeDelta::Seconds(1) / GetCurrentConfiguredFramerate();
    });
  }
}
void FrameGeneratorCapturer::InsertFrame() {
  if (sending_) {
    FrameGeneratorInterface::VideoFrameData frame_data =
        frame_generator_->NextFrame();
    int decimation =
        std::round(static_cast<double>(source_fps_) / target_capture_fps_);
    for (int i = 1; i < decimation; ++i)
      frame_data = frame_generator_->NextFrame();

    VideoFrame frame = VideoFrame::Builder()
                           .set_video_frame_buffer(frame_data.buffer)
                           .set_rotation(fake_rotation_)
                           .set_timestamp_us(clock_->TimeInMicroseconds())
                           .set_ntp_time_ms(clock_->CurrentNtpInMilliseconds())
                           .set_update_rect(frame_data.update_rect)
                           .set_color_space(fake_color_space_)
                           .build();
    if (first_frame_capture_time_ == -1) {
      first_frame_capture_time_ = frame.ntp_time_ms();
    }
    TestVideoCapturer::OnFrame(frame);
  }
}
void TestVideoCapturer::OnFrame(const VideoFrame& original_frame) {
    //帧数据 投递给VideoStreamEncoder
    broadcaster_.OnFrame(frame);
}

encode

视频怎么从Capturer传递给encoder？SendVideoStream::SendVideoStream函数的下半部分有交代。

//third_party/webrtc/test/scenario/video_stream.cc
SendVideoStream::SendVideoStream(CallClient* sender,
                                 VideoStreamConfig config,
                                 Transport* send_transport,
                                 VideoFrameMatcher* matcher){
  video_capturer_ = std::make_unique<FrameGeneratorCapturer>(
      sender_->clock_, CreateFrameGenerator(sender_->clock_, config.source),
      config.source.framerate,
      *sender->time_controller_->GetTaskQueueFactory());
  video_capturer_->Init();                                 
//...
  bitrate_allocator_factory_ = CreateBuiltinVideoBitrateAllocatorFactory();
  VideoSendStream::Config send_config =
      CreateVideoSendStreamConfig(config, ssrcs_, rtx_ssrcs_, send_transport);
   // analysis
  send_config.encoder_settings.encoder_factory = encoder_factory_.get();
  send_config.encoder_settings.bitrate_allocator_factory =
      bitrate_allocator_factory_.get();
  //sender_ type defined in third_party/webrtc/test/scenario/call_client.cc
//在第四部分分析 编码到传输
  sender_->SendTask([&] {
    if (config.stream.fec_controller_factory) {
      send_stream_ = sender_->call_->CreateVideoSendStream(
          std::move(send_config), std::move(encoder_config),
          config.stream.fec_controller_factory->CreateFecController());
    } else {
      send_stream_ = sender_->call_->CreateVideoSendStream(
          std::move(send_config), std::move(encoder_config));
    }

    if (matcher->Active()) {
      frame_tap_ = std::make_unique<ForwardingCapturedFrameTap>(
          sender_->clock_, matcher, video_capturer_.get());
      send_stream_->SetSource(frame_tap_.get(),
                              config.encoder.degradation_preference);
    } else {
      send_stream_->SetSource(video_capturer_.get(),
                              config.encoder.degradation_preference);
    }
  });                          
}

send_config.encoder_settings.encoder_factory = encoder_factory_.get()，视频编码器工厂。
创建VideoSendStream，其中根据encoder_settings创建编码器。

//third_party/webrtc/call/call.cc
webrtc::VideoSendStream* Call::CreateVideoSendStream(
    webrtc::VideoSendStream::Config config,
    VideoEncoderConfig encoder_config,
    std::unique_ptr<FecController> fec_controller) {
  VideoSendStream* send_stream = new VideoSendStream(
      clock_, num_cpu_cores_, module_process_thread_->process_thread(),
      task_queue_factory_, call_stats_->AsRtcpRttStats(), transport_send_ptr_,
      bitrate_allocator_.get(), video_send_delay_stats_.get(), event_log_,
      std::move(config), std::move(encoder_config), suspended_video_send_ssrcs_,
      suspended_video_payload_states_, std::move(fec_controller));        
    }

VideoSendStream::VideoSendStream(
    Clock* clock,
    int num_cpu_cores,
    ProcessThread* module_process_thread,
    TaskQueueFactory* task_queue_factory,
    RtcpRttStats* call_stats,
    RtpTransportControllerSendInterface* transport,
    BitrateAllocatorInterface* bitrate_allocator,
    SendDelayStats* send_delay_stats,
    RtcEventLog* event_log,
    VideoSendStream::Config config,
    VideoEncoderConfig encoder_config,
    const std::map<uint32_t, RtpState>& suspended_ssrcs,
    const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
    std::unique_ptr<FecController> fec_controller)
    : worker_queue_(transport->GetWorkerQueue()),
      stats_proxy_(clock, config, encoder_config.content_type),
      config_(std::move(config)),
      content_type_(encoder_config.content_type) {
  video_stream_encoder_ =
      CreateVideoStreamEncoder(clock, task_queue_factory, num_cpu_cores,
                               &stats_proxy_, config_.encoder_settings);
  worker_queue_->PostTask(ToQueuedTask(
      [this, clock, call_stats, transport, bitrate_allocator, send_delay_stats,
       event_log, &suspended_ssrcs, &encoder_config, &suspended_payload_states,
       &fec_controller]() {
        send_stream_.reset(new VideoSendStreamImpl(
            clock, &stats_proxy_, worker_queue_, call_stats, transport,
            bitrate_allocator, send_delay_stats, video_stream_encoder_.get(),
            event_log, &config_, encoder_config.max_bitrate_bps,
            encoder_config.bitrate_priority, suspended_ssrcs,
            suspended_payload_states, encoder_config.content_type,
            std::move(fec_controller)));
      },
      [this]() { thread_sync_event_.Set(); }));                
    //VideoSendStreamImpl向video_stream_encoder_注册自己为sink，获取编码后的数据
    // ?                           
      ReconfigureVideoEncoder(std::move(encoder_config));
      }

//third_party/webrtc/api/video/video_stream_encoder_create.cc
std::unique_ptr<VideoStreamEncoderInterface> CreateVideoStreamEncoder(
    Clock* clock,
    TaskQueueFactory* task_queue_factory,
    uint32_t number_of_cores,
    VideoStreamEncoderObserver* encoder_stats_observer,
    const VideoStreamEncoderSettings& settings) {
  return std::make_unique<VideoStreamEncoder>(
      clock, number_of_cores, encoder_stats_observer, settings,
      std::make_unique<OveruseFrameDetector>(encoder_stats_observer),
      task_queue_factory);
}
VideoStreamEncoder::VideoStreamEncoder(
    Clock* clock,
    uint32_t number_of_cores,
    VideoStreamEncoderObserver* encoder_stats_observer,
    const VideoStreamEncoderSettings& settings,
    std::unique_ptr<OveruseFrameDetector> overuse_detector,
    TaskQueueFactory* task_queue_factory)
    : shutdown_event_(true /* manual_reset */, false),
      number_of_cores_(number_of_cores),
      quality_scaling_experiment_enabled_(QualityScalingExperiment::Enabled()),
      sink_(nullptr),
      settings_(settings){
     // settings_的作用，包含编码器工厂
      }

创建并配置编码器：

//third_party/webrtc/video/video_send_stream.cc
void VideoSendStream::ReconfigureVideoEncoder(VideoEncoderConfig config) {
  // TODO(perkj): Some test cases in VideoSendStreamTest call
  // ReconfigureVideoEncoder from the network thread.
  // RTC_DCHECK_RUN_ON(&thread_checker_);
  RTC_DCHECK(content_type_ == config.content_type);
  video_stream_encoder_->ConfigureEncoder(
      std::move(config),
      config_.rtp.max_packet_size - CalculateMaxHeaderSize(config_.rtp));
}
//third_party/webrtc/video/video_stream_encoder.cc
void VideoStreamEncoder::ConfigureEncoder(VideoEncoderConfig config,
                                          size_t max_data_payload_length) {
  encoder_queue_.PostTask(
      [this, config = std::move(config), max_data_payload_length]() mutable {
        RTC_DCHECK_RUN_ON(&encoder_queue_);
        RTC_DCHECK(sink_);
        RTC_LOG(LS_INFO) << "ConfigureEncoder requested.";

        pending_encoder_creation_ =
            (!encoder_ || encoder_config_.video_format != config.video_format ||
             max_data_payload_length_ != max_data_payload_length);
        encoder_config_ = std::move(config);
        max_data_payload_length_ = max_data_payload_length;
        pending_encoder_reconfiguration_ = true;

        // Reconfigure the encoder now if the encoder has an internal source or
        // if the frame resolution is known. Otherwise, the reconfiguration is
        // deferred until the next frame to minimize the number of
        // reconfigurations. The codec configuration depends on incoming video
        // frame size.
        if (last_frame_info_) {
          ReconfigureEncoder();
        } else {
          codec_info_ = settings_.encoder_factory->QueryVideoEncoder(
              encoder_config_.video_format);
          if (HasInternalSource()) {
            last_frame_info_ = VideoFrameInfo(kDefaultInputPixelsWidth,
                                              kDefaultInputPixelsHeight, false);
            ReconfigureEncoder();
          }
        }
      });
}

创建编码器：

//third_party/webrtc/video/video_stream_encoder.cc
void VideoStreamEncoder::ReconfigureEncoder() {
  if (pending_encoder_creation_) {
    // Destroy existing encoder instance before creating a new one. Otherwise
    // attempt to create another instance will fail if encoder factory
    // supports only single instance of encoder of given type.
    encoder_.reset();

    encoder_ = settings_.encoder_factory->CreateVideoEncoder(
        encoder_config_.video_format);
    // TODO(nisse): What to do if creating the encoder fails? Crash,
    // or just discard incoming frames?
    RTC_CHECK(encoder_);

    if (encoder_selector_) {
      encoder_selector_->OnCurrentEncoder(encoder_config_.video_format);
    }

    encoder_->SetFecControllerOverride(fec_controller_override_);

    codec_info_ = settings_.encoder_factory->QueryVideoEncoder(
        encoder_config_.video_format);

    encoder_reset_required = true;
  }
}

假设仿真的时候，采用的是test::FakeEncoder。编码输出的数据，按照current_rate_settings_中的配置信息生成。

int32_t FakeEncoder::Encode(const VideoFrame& input_image,
                            const std::vector<VideoFrameType>* frame_types) {
	 rates = current_rate_settings_;
 	 FrameInfo frame_info =
      NextFrame(frame_types, keyframe, num_simulcast_streams, rates.bitrate,
                simulcast_streams, static_cast<int>(rates.framerate_fps + 0.5));	                            
}

码率的设置接口。SetRates在VideoStreamEncoder中被调用。

void FakeEncoder::SetRates(const RateControlParameters& parameters) {
  rtc::CritScope cs(&crit_sect_);
  SetRatesLocked(parameters);
}

void FakeEncoder::SetRatesLocked(const RateControlParameters& parameters) {
  current_rate_settings_ = parameters;
  int allocated_bitrate_kbps = parameters.bitrate.get_sum_kbps();

  // Scale bitrate allocation to not exceed the given max target bitrate.
  if (max_target_bitrate_kbps_ > 0 &&
      allocated_bitrate_kbps > max_target_bitrate_kbps_) {
    for (uint8_t spatial_idx = 0; spatial_idx < kMaxSpatialLayers;
         ++spatial_idx) {
      for (uint8_t temporal_idx = 0; temporal_idx < kMaxTemporalStreams;
           ++temporal_idx) {
        if (current_rate_settings_.bitrate.HasBitrate(spatial_idx,
                                                      temporal_idx)) {
          uint32_t bitrate = current_rate_settings_.bitrate.GetBitrate(
              spatial_idx, temporal_idx);
          bitrate = static_cast<uint32_t>(
              (bitrate * int64_t{max_target_bitrate_kbps_}) /
              allocated_bitrate_kbps);
          current_rate_settings_.bitrate.SetBitrate(spatial_idx, temporal_idx,
                                                    bitrate);
        }
      }
    }
  }
}

Adjust the bitrate of encoder

上一节中给出了编码器码率的设置接口。这里分析，从拥塞控制器获取速率，到配置编码器速率的流程。RtpTransportControllerSend中会从拥塞控制器中获取速率信息。更改编码器速率，有两个出发点，一个就是拥塞控制器速率发生变化的时候，另一个就是周期性地从拥塞控制器获取速率。

//third_party/webrtc/call/rtp_transport_controller_send.cc
//one 
void RtpTransportControllerSend::PostUpdates(NetworkControlUpdate update) {
  if (update.target_rate) {
    control_handler_->SetTargetRate(*update.target_rate);
    UpdateControlState();
  }
}
//two   
//constexpr TimeDelta kPacerQueueUpdateInterval = TimeDelta::Millis(25);
void RtpTransportControllerSend::StartProcessPeriodicTasks() {
  if (!pacer_queue_update_task_.Running()) {
    pacer_queue_update_task_ = RepeatingTaskHandle::DelayedStart(
        task_queue_.Get(), kPacerQueueUpdateInterval, [this]() {
          RTC_DCHECK_RUN_ON(&task_queue_);
          TimeDelta expected_queue_time = pacer()->ExpectedQueueTime();
          control_handler_->SetPacerQueue(expected_queue_time);
          UpdateControlState();
          return kPacerQueueUpdateInterval;
        });
  }
}

从UpdateControlState走起。

void RtpTransportControllerSend::UpdateControlState() {
  absl::optional<TargetTransferRate> update = control_handler_->GetUpdate();
  if (!update)
    return;
  retransmission_rate_limiter_.SetMaxRate(update->target_rate.bps());
  observer_->OnTargetTransferRate(*update);
}

//third_party/webrtc/call/call.cc
void Call::OnTargetTransferRate(TargetTransferRate msg) {
    uint32_t target_bitrate_bps = msg.target_rate.bps();
  // For controlling the rate of feedback messages.
    receive_side_cc_.OnBitrateChanged(target_bitrate_bps);
    bitrate_allocator_->OnNetworkEstimateChanged(msg);
    
    /*
  worker_thread_->PostTask(
      ToQueuedTask(task_safety_, [this, target_bitrate_bps]() {
        RTC_DCHECK_RUN_ON(worker_thread_);
        last_bandwidth_bps_ = target_bitrate_bps;

        // Ignore updates if bitrate is zero (the aggregate network state is
        // down) or if we're not sending video.
        if (target_bitrate_bps == 0 || video_send_streams_.empty()) {
          estimated_send_bitrate_kbps_counter_.ProcessAndPause();
          pacer_bitrate_kbps_counter_.ProcessAndPause();
          return;
        }

        estimated_send_bitrate_kbps_counter_.Add(target_bitrate_bps / 1000);
        // Pacer bitrate may be higher than bitrate estimate if enforcing min
        // bitrate.
        uint32_t pacer_bitrate_bps =
            std::max(target_bitrate_bps, min_allocated_send_bitrate_bps_);
        pacer_bitrate_kbps_counter_.Add(pacer_bitrate_bps / 1000);
      }));    
    */
}


//third_party/webrtc/call/bitrate_allocator.cc
void BitrateAllocator::OnNetworkEstimateChanged(TargetTransferRate msg) {
    uint32_t protection_bitrate = config.observer->OnBitrateUpdated(update);
}
//third_party/webrtc/video/video_send_stream_impl.cc
uint32_t VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update) {
  rtp_video_sender_->OnBitrateUpdated(update, stats_proxy_->GetSendFrameRate());
  encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps();
  const uint32_t protection_bitrate_bps =
      rtp_video_sender_->GetProtectionBitrateBps();
  DataRate link_allocation = DataRate::Zero();
  if (encoder_target_rate_bps_ > protection_bitrate_bps) {
    link_allocation =
        DataRate::BitsPerSec(encoder_target_rate_bps_ - protection_bitrate_bps);
  }
  DataRate overhead =
      update.target_bitrate - DataRate::BitsPerSec(encoder_target_rate_bps_);
  DataRate encoder_stable_target_rate = update.stable_target_bitrate;
  if (encoder_stable_target_rate > overhead) {
    encoder_stable_target_rate = encoder_stable_target_rate - overhead;
  } else {
    encoder_stable_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_);
  }

  encoder_target_rate_bps_ =
      std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);

  encoder_stable_target_rate =
      std::min(DataRate::BitsPerSec(encoder_max_bitrate_bps_),
               encoder_stable_target_rate);

  DataRate encoder_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_);
  link_allocation = std::max(encoder_target_rate, link_allocation);
  video_stream_encoder_->OnBitrateUpdated(
      encoder_target_rate, encoder_stable_target_rate, link_allocation,
      rtc::dchecked_cast<uint8_t>(update.packet_loss_ratio * 256),
      update.round_trip_time.ms(), update.cwnd_reduce_ratio);    
}

//third_party/webrtc/video/video_stream_encoder.cc
void VideoStreamEncoder::OnBitrateUpdated(DataRate target_bitrate,
                                          DataRate stable_target_bitrate,
                                          DataRate link_allocation,
                                          uint8_t fraction_lost,
                                          int64_t round_trip_time_ms,
                                          double cwnd_reduce_ratio) {
  EncoderRateSettings new_rate_settings{
      VideoBitrateAllocation(), static_cast<double>(framerate_fps),
      link_allocation, target_bitrate, stable_target_bitrate};
  SetEncoderRates(UpdateBitrateAllocationAndNotifyObserver(new_rate_settings));                                              
                                          }


//third_party/webrtc/video/video_stream_encoder.cc
void VideoStreamEncoder::SetEncoderRates(
    const EncoderRateSettings& rate_settings) {
        encoder_->SetRates(rate_settings.rate_control);        
    }

VideoSendStreamImpl注册到BitrateAllocator。

EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage(
    const EncodedImage& encoded_image,
    const CodecSpecificInfo* codec_specific_info,
    const RTPFragmentationHeader* fragmentation) {
  auto enable_padding_task = [this]() {
    if (disable_padding_) {
      RTC_DCHECK_RUN_ON(worker_queue_);
      disable_padding_ = false;
      // To ensure that padding bitrate is propagated to the bitrate allocator.
      SignalEncoderActive();
    }
  };
  if (!worker_queue_->IsCurrent()) {
    worker_queue_->PostTask(enable_padding_task);
  } else {
    enable_padding_task();
  }
  EncodedImageCallback::Result result(EncodedImageCallback::Result::OK);
  result = rtp_video_sender_->OnEncodedImage(encoded_image, codec_specific_info,
                                             fragmentation);  
    }
void VideoSendStreamImpl::SignalEncoderActive() {
  RTC_DCHECK_RUN_ON(worker_queue_);
  if (rtp_video_sender_->IsActive()) {
    RTC_LOG(LS_INFO) << "SignalEncoderActive, Encoder is active.";
    bitrate_allocator_->AddObserver(this, GetAllocationConfig());
  }
}
void VideoSendStreamImpl::StartupVideoSendStream() {
    bitrate_allocator_->AddObserver(this, GetAllocationConfig());
}
void BitrateAllocator::AddObserver(BitrateAllocatorObserver* observer,
                                   MediaStreamAllocationConfig config) {
  auto it = absl::c_find_if(
      allocatable_tracks_,
      [observer](const auto& config) { return config.observer == observer; });
  // Update settings if the observer already exists, create a new one otherwise.
  if (it != allocatable_tracks_.end()) {
    it->config = config;
  } else {
    allocatable_tracks_.push_back(AllocatableTrack(observer, config));
  }                                       
}

From encoder to rtp sender

//third_party/webrtc/test/scenario/video_stream.cc
SendVideoStream::SendVideoStream(CallClient* sender,
                                 VideoStreamConfig config,
                                 Transport* send_transport,
                                 VideoFrameMatcher* matcher){
    if (config.stream.fec_controller_factory) {
      send_stream_ = sender_->call_->CreateVideoSendStream(
          std::move(send_config), std::move(encoder_config),
          config.stream.fec_controller_factory->CreateFecController());
    } else {
      send_stream_ = sender_->call_->CreateVideoSendStream(
          std::move(send_config), std::move(encoder_config));
    }
    send_stream_->SetSource(video_capturer_.get(),
                              config.encoder.degradation_preference);    
}

FrameGeneratorCapturer产生数据，传递到encoder处理，encoder将数据传递到sink( VideoSendStreamImpl)。

void VideoSendStream::SetSource(
    rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
    const DegradationPreference& degradation_preference) {
  RTC_DCHECK_RUN_ON(&thread_checker_);
  video_stream_encoder_->SetSource(source, degradation_preference);
}
int32_t FakeEncoder::Encode(const VideoFrame& input_image,
                            const std::vector<VideoFrameType>* frame_types) {
    if (callback->OnEncodedImage(encoded, &codec_specific, fragmentation.get())
            .error != EncodedImageCallback::Result::OK) {
      return -1;
    }                                
                            }
EncodedImageCallback::Result VideoStreamEncoder::OnEncodedImage(
    const EncodedImage& encoded_image,
    const CodecSpecificInfo* codec_specific_info,
    const RTPFragmentationHeader* fragmentation){
  EncodedImageCallback::Result result = sink_->OnEncodedImage(
      image_copy, codec_specific_info,
      fragmentation_copy ? fragmentation_copy.get() : fragmentation);        
}
EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage(
    const EncodedImage& encoded_image,
    const CodecSpecificInfo* codec_specific_info,
    const RTPFragmentationHeader* fragmentation){
    result = rtp_video_sender_->OnEncodedImage(encoded_image, codec_specific_info,
                                             fragmentation);        
 }

接下来就要分析rtp_video_sender_。

EncodedImageCallback::Result RtpVideoSender::OnEncodedImage(
    const EncodedImage& encoded_image,
    const CodecSpecificInfo* codec_specific_info,
    const RTPFragmentationHeader* fragmentation) {
  uint32_t rtp_timestamp =
      encoded_image.Timestamp() +
      rtp_streams_[stream_index].rtp_rtcp->StartTimestamp();   
  bool send_result = rtp_streams_[stream_index].sender_video->SendEncodedImage(
      rtp_config_.payload_type, codec_type_, rtp_timestamp, encoded_image,
      fragmentation,
      params_[stream_index].GetRtpVideoHeader(
          encoded_image, codec_specific_info, shared_frame_id_),
      expected_retransmission_time_ms);        
    }        
bool RTPSenderVideo::SendEncodedImage(
    int payload_type,
    absl::optional<VideoCodecType> codec_type,
    uint32_t rtp_timestamp,
    const EncodedImage& encoded_image,
    const RTPFragmentationHeader* fragmentation,
    RTPVideoHeader video_header,
    absl::optional<int64_t> expected_retransmission_time_ms) {
  return SendVideo(payload_type, codec_type, rtp_timestamp,
                   encoded_image.capture_time_ms_, encoded_image, fragmentation,
                   video_header, expected_retransmission_time_ms);        
    }
bool RTPSenderVideo::SendVideo(
    int payload_type,
    absl::optional<VideoCodecType> codec_type,
    uint32_t rtp_timestamp,
    int64_t capture_time_ms,
    rtc::ArrayView<const uint8_t> payload,
    const RTPFragmentationHeader* fragmentation,
    RTPVideoHeader video_header,
    absl::optional<int64_t> expected_retransmission_time_ms) {
  AddRtpHeaderExtensions(video_header, absolute_capture_time,
                         /*first_packet=*/true, /*last_packet=*/true,
                         single_packet.get());
  AddRtpHeaderExtensions(video_header, absolute_capture_time,
                         /*first_packet=*/true, /*last_packet=*/false,
                         first_packet.get());
  AddRtpHeaderExtensions(video_header, absolute_capture_time,
                         /*first_packet=*/false, /*last_packet=*/false,
                         middle_packet.get());
  AddRtpHeaderExtensions(video_header, absolute_capture_time,
                         /*first_packet=*/false, /*last_packet=*/true,
                         last_packet.get());
  std::unique_ptr<RtpPacketizer> packetizer = RtpPacketizer::Create(
      codec_type, payload, limits, video_header, fragmentation);
  const size_t num_packets = packetizer->NumPackets();   
/*
    // No FEC protection for upper temporal layers, if used.
    if (fec_type_.has_value() &&
        (temporal_id == 0 || temporal_id == kNoTemporalIdx)) {
      if (fec_generator_) {
        fec_generator_->AddPacketAndGenerateFec(*packet);
      } else {
        // Deferred FEC generation, just mark packet.
        packet->set_fec_protect_packet(true);
      }
    }

    if (red_enabled()) {
      std::unique_ptr<RtpPacketToSend> red_packet(new RtpPacketToSend(*packet));
      BuildRedPayload(*packet, red_packet.get());
      red_packet->SetPayloadType(*red_payload_type_);
      red_packet->set_is_red(true);

      // Send |red_packet| instead of |packet| for allocated sequence number.
      red_packet->set_packet_type(RtpPacketMediaType::kVideo);
      red_packet->set_allow_retransmission(packet->allow_retransmission());
      rtp_packets.emplace_back(std::move(red_packet));
    } else {
      packet->set_packet_type(RtpPacketMediaType::kVideo);
      rtp_packets.emplace_back(std::move(packet));
    }
*/  
LogAndSendToNetwork(std::move(rtp_packets), unpacketized_payload_size);
    }
void RTPSenderVideo::LogAndSendToNetwork(
    std::vector<std::unique_ptr<RtpPacketToSend>> packets,
    size_t unpacketized_payload_size) {
  {
    MutexLock lock(&stats_mutex_);
    size_t packetized_payload_size = 0;
    for (const auto& packet : packets) {
      if (*packet->packet_type() == RtpPacketMediaType::kVideo) {
        video_bitrate_.Update(packet->size(), now_ms);
        packetized_payload_size += packet->payload_size();
      }
    }
    // AV1 packetizer may produce less packetized bytes than unpacketized.
    if (packetized_payload_size >= unpacketized_payload_size) {
      packetization_overhead_bitrate_.Update(
          packetized_payload_size - unpacketized_payload_size,
          clock_->TimeInMilliseconds());
    }
  }

  rtp_sender_->EnqueuePackets(std::move(packets));        
    }
void RTPSender::EnqueuePackets(
    std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
  RTC_DCHECK(!packets.empty());
  int64_t now_ms = clock_->TimeInMilliseconds();
  for (auto& packet : packets) {
    RTC_DCHECK(packet);
    RTC_CHECK(packet->packet_type().has_value())
        << "Packet type must be set before sending.";
    if (packet->capture_time_ms() <= 0) {
      packet->set_capture_time_ms(now_ms);
    }
  }

  paced_sender_->EnqueuePackets(std::move(packets));
}

分析到最后，发现数据包不是立马发送到网络中，而是将数据交给了pacer处理。

Transmission

还是从仿真器的初始化看起。回到文章开始的地方。

//third_party/webrtc/test/scenario/call_client.cc
CallClient::CallClient(
    TimeController* time_controller,
    std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
    CallClientConfig config)
    : time_controller_(time_controller),
      clock_(time_controller->GetClock()),
      log_writer_factory_(std::move(log_writer_factory)),
      network_controller_factory_(log_writer_factory_.get(), config.transport),
      header_parser_(RtpHeaderParser::CreateForTest()),
      task_queue_(time_controller->GetTaskQueueFactory()->CreateTaskQueue(
          "CallClient",
          TaskQueueFactory::Priority::NORMAL)) {
    call_.reset(CreateCall(time_controller_, event_log_.get(), config,
                           &network_controller_factory_,
                           fake_audio_setup_.audio_state, module_thread_));
    transport_ = std::make_unique<NetworkNodeTransport>(clock_, call_.get());              
          }

Call* CreateCall(TimeController* time_controller,
                 RtcEventLog* event_log,
                 CallClientConfig config,
                 LoggingNetworkControllerFactory* network_controller_factory,
                 rtc::scoped_refptr<AudioState> audio_state,
                 rtc::scoped_refptr<SharedModuleThread> call_thread) {
  CallConfig call_config(event_log);
  call_config.bitrate_config.max_bitrate_bps =
      config.transport.rates.max_rate.bps_or(-1);
  call_config.bitrate_config.min_bitrate_bps =
      config.transport.rates.min_rate.bps();
  call_config.bitrate_config.start_bitrate_bps =
      config.transport.rates.start_rate.bps();
  call_config.task_queue_factory = time_controller->GetTaskQueueFactory();
  call_config.network_controller_factory = network_controller_factory;
  call_config.audio_state = audio_state;
  call_config.trials = config.field_trials;
  return Call::Create(call_config, time_controller->GetClock(),
                      std::move(call_thread),
                      time_controller->CreateProcessThread("Pacer"));
}
//third_party/webrtc/call/call.cc
Call* Call::Create(const Call::Config& config,
                   Clock* clock,
                   rtc::scoped_refptr<SharedModuleThread> call_thread,
                   std::unique_ptr<ProcessThread> pacer_thread) {
  RTC_DCHECK(config.task_queue_factory);
  return new internal::Call(
      clock, config,
      std::make_unique<RtpTransportControllerSend>(
          clock, config.event_log, config.network_state_predictor_factory,
          config.network_controller_factory, config.bitrate_config,
          std::move(pacer_thread), config.task_queue_factory, config.trials),
      std::move(call_thread), config.task_queue_factory);
}

注册发送数据的transport(sender->transport_.get())。

VideoStreamPair* Scenario::CreateVideoStream(
    std::pair<CallClient*, CallClient*> clients,
    VideoStreamConfig config) {
  video_streams_.emplace_back(
      new VideoStreamPair(clients.first, clients.second, config));
  return video_streams_.back().get();
}
VideoStreamPair::VideoStreamPair(CallClient* sender,
                                 CallClient* receiver,
                                 VideoStreamConfig config)
    : config_(config),
      matcher_(config.hooks.frame_pair_handlers),
      send_stream_(sender, config, sender->transport_.get(), &matcher_),
      receive_stream_(receiver,
                      config,
                      &send_stream_,
                      /*chosen_stream=*/0,
                      receiver->transport_.get(),
                      &matcher_) {}
//third_party/webrtc/test/scenario/video_stream.cc
SendVideoStream::SendVideoStream(CallClient* sender,
                                 VideoStreamConfig config,
                                 Transport* send_transport,
                                 VideoFrameMatcher* matcher){
  VideoSendStream::Config send_config =
      CreateVideoSendStreamConfig(config, ssrcs_, rtx_ssrcs_, send_transport);
      send_stream_ = sender_->call_->CreateVideoSendStream(
          std::move(send_config), std::move(encoder_config));      
}
//send_transport
VideoSendStreamImpl::VideoSendStreamImpl():
rtp_video_sender_(transport_->CreateRtpVideoSender(suspended_ssrcs,
                  suspended_payload_states,
                  config_->rtp,
                  config_->rtcp_report_interval_ms,
                  config_->send_transport,
                  CreateObservers(call_stats,
                  &encoder_feedback_,
                  stats_proxy_,
                  send_delay_stats){
    
}
RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender(
    std::map<uint32_t, RtpState> suspended_ssrcs,
    const std::map<uint32_t, RtpPayloadState>& states,
    const RtpConfig& rtp_config,
    int rtcp_report_interval_ms,
    Transport* send_transport,
    const RtpSenderObservers& observers,
    RtcEventLog* event_log,
    std::unique_ptr<FecController> fec_controller,
    const RtpSenderFrameEncryptionConfig& frame_encryption_config,
    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
  video_rtp_senders_.push_back(std::make_unique<RtpVideoSender>(
      clock_, suspended_ssrcs, states, rtp_config, rtcp_report_interval_ms,
      send_transport, observers,
      // TODO(holmer): Remove this circular dependency by injecting
      // the parts of RtpTransportControllerSendInterface that are really used.
      this, event_log, &retransmission_rate_limiter_, std::move(fec_controller),
      frame_encryption_config.frame_encryptor,
      frame_encryption_config.crypto_options, std::move(frame_transformer)));
  return video_rtp_senders_.back().get();        
    }

RtpVideoSender::RtpVideoSender(
    Clock* clock,
    std::map<uint32_t, RtpState> suspended_ssrcs,
    const std::map<uint32_t, RtpPayloadState>& states,
    const RtpConfig& rtp_config,
    int rtcp_report_interval_ms,
    Transport* send_transport,
    const RtpSenderObservers& observers,
    RtpTransportControllerSendInterface* transport,
    RtcEventLog* event_log,
    RateLimiter* retransmission_limiter,
    std::unique_ptr<FecController> fec_controller,
    FrameEncryptorInterface* frame_encryptor,
    const CryptoOptions& crypto_options,
    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer)
    : send_side_bwe_with_overhead_(absl::StartsWith(
          field_trials_.Lookup("WebRTC-SendSideBwe-WithOverhead"),
          "Enabled")),
      account_for_packetization_overhead_(!absl::StartsWith(
          field_trials_.Lookup("WebRTC-SubtractPacketizationOverhead"),
          "Disabled")),
      use_early_loss_detection_(!absl::StartsWith(
          field_trials_.Lookup("WebRTC-UseEarlyLossDetection"),
          "Disabled")),
      has_packet_feedback_(TransportSeqNumExtensionConfigured(rtp_config)),
      use_deferred_fec_(
          absl::StartsWith(field_trials_.Lookup("WebRTC-DeferredFecGeneration"),
                           "Enabled")),
      active_(false),
      module_process_thread_(nullptr),
      suspended_ssrcs_(std::move(suspended_ssrcs)),
      fec_controller_(std::move(fec_controller)),
      fec_allowed_(true),
      rtp_streams_(CreateRtpStreamSenders(clock,
                                          rtp_config,
                                          observers,
                                          rtcp_report_interval_ms,
                                          send_transport,
                                          transport->GetBandwidthObserver(),
                                          transport,
                                          suspended_ssrcs_,
                                          event_log,
                                          retransmission_limiter,
                                          frame_encryptor,
                                          crypto_options,
                                          std::move(frame_transformer),
                                          use_deferred_fec_,
                                          field_trials_)),
      rtp_config_(rtp_config),
      codec_type_(GetVideoCodecType(rtp_config)),
      transport_(transport),
      transport_overhead_bytes_per_packet_(0),
      encoder_target_rate_bps_(0),
      frame_counts_(rtp_config.ssrcs.size()),
      frame_count_observer_(observers.frame_count_observer) {
  for (const RtpStreamSender& stream : rtp_streams_) {
    constexpr bool remb_candidate = true;
    transport->packet_router()->AddSendRtpModule(stream.rtp_rtcp.get(),
                                                 remb_candidate);
  }
}

void PacketRouter::AddSendRtpModule(RtpRtcpInterface* rtp_module,
                                    bool remb_candidate) {}
void PacketRouter::AddSendRtpModuleToMap(RtpRtcpInterface* rtp_module,
                                         uint32_t ssrc) {
  RTC_DCHECK(send_modules_map_.find(ssrc) == send_modules_map_.end());
  // Always keep the audio modules at the back of the list, so that when we
  // iterate over the modules in order to find one that can send padding we
  // will prioritize video. This is important to make sure they are counted
  // into the bandwidth estimate properly.
  if (rtp_module->IsAudioConfigured()) {
    send_modules_list_.push_back(rtp_module);
  } else {
    send_modules_list_.push_front(rtp_module);
  }
  send_modules_map_[ssrc] = rtp_module;
}

从transport发送数据。

void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
                              const PacedPacketInfo& cluster_info) {
  uint32_t ssrc = packet->Ssrc();
  auto kv = send_modules_map_.find(ssrc);                                 
  RtpRtcpInterface* rtp_module = kv->second;
  if (!rtp_module->TrySendPacket(packet.get(), cluster_info)) {
    RTC_LOG(LS_WARNING) << "Failed to send packet, rejected by RTP module.";
    return;
  }                                  
}
bool ModuleRtpRtcpImpl2::TrySendPacket(RtpPacketToSend* packet,
                                       const PacedPacketInfo& pacing_info) {
  RTC_DCHECK(rtp_sender_);
  // TODO(sprang): Consider if we can remove this check.
  if (!rtp_sender_->packet_generator.SendingMedia()) {
    return false;
  }
  rtp_sender_->packet_sender.SendPacket(packet, pacing_info);
  return true;
}
void RtpSenderEgress::SendPacket(RtpPacketToSend* packet,
                                 const PacedPacketInfo& pacing_info) {
 const bool send_success = SendPacketToNetwork(*packet, options, pacing_info);                                    
}
bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
                                          const PacketOptions& options,
                                          const PacedPacketInfo& pacing_info) {
  if (transport_) {
    bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
                     ? static_cast<int>(packet.size())
                     : -1;
    if (event_log_ && bytes_sent > 0) {
      event_log_->Log(std::make_unique<RtcEventRtpPacketOutgoing>(
          packet, pacing_info.probe_cluster_id));
    }
  }                                              
}

ModuleRtpRtcpImpl2中有rtcp_sender_和rtp_sender_。数据均是从transport发送。
仿真器中实现的transport。

bool NetworkNodeTransport::SendRtp(const uint8_t* packet,
                                   size_t length,
                                   const PacketOptions& options) {
  int64_t send_time_ms = sender_clock_->TimeInMilliseconds();
  rtc::SentPacket sent_packet;
  sent_packet.packet_id = options.packet_id;
  sent_packet.info.included_in_feedback = options.included_in_feedback;
  sent_packet.info.included_in_allocation = options.included_in_allocation;
  sent_packet.send_time_ms = send_time_ms;
  sent_packet.info.packet_size_bytes = length;
  sent_packet.info.packet_type = rtc::PacketType::kData;
  sender_call_->OnSentPacket(sent_packet);

  MutexLock lock(&mutex_);
  if (!endpoint_)
    return false;
  rtc::CopyOnWriteBuffer buffer(packet, length);
  endpoint_->SendPacket(local_address_, remote_address_, buffer,
                        packet_overhead_.bytes());
  return true;
}

bool NetworkNodeTransport::SendRtcp(const uint8_t* packet, size_t length) {
  rtc::CopyOnWriteBuffer buffer(packet, length);
  MutexLock lock(&mutex_);
  if (!endpoint_)
    return false;
  endpoint_->SendPacket(local_address_, remote_address_, buffer,
                        packet_overhead_.bytes());
  return true;
}

Incoming data processing

只分析在仿真器中的数据接收流程。

CallClientPair* Scenario::CreateRoutes(
    CallClient* first,
    std::vector<EmulatedNetworkNode*> send_link,
    DataSize first_overhead,
    CallClient* second,
    std::vector<EmulatedNetworkNode*> return_link,
    DataSize second_overhead) {
  CallClientPair* client_pair = new CallClientPair(first, second);
  ChangeRoute(client_pair->forward(), send_link, first_overhead);
  ChangeRoute(client_pair->reverse(), return_link, second_overhead);
  client_pairs_.emplace_back(client_pair);
  return client_pair;
}
void Scenario::ChangeRoute(std::pair<CallClient*, CallClient*> clients,
                           std::vector<EmulatedNetworkNode*> over_nodes,
                           DataSize overhead) {
  EmulatedRoute* route = network_manager_.CreateRoute(over_nodes);
  uint16_t port = clients.second->Bind(route->to);
  auto addr = rtc::SocketAddress(route->to->GetPeerLocalAddress(), port);
  clients.first->transport_->Connect(route->from, addr, overhead);
}
int16_t CallClient::Bind(EmulatedEndpoint* endpoint) {
  uint16_t port = endpoint->BindReceiver(0, this).value();
  endpoints_.push_back({endpoint, port});
  return port;
}

endpoint接收到数据包后，会调用CallClient中的数据接收接口。

void CallClient::OnPacketReceived(EmulatedIpPacket packet) {
  MediaType media_type = MediaType::ANY;
  if (!RtpHeaderParser::IsRtcp(packet.cdata(), packet.data.size())) {
    auto ssrc = RtpHeaderParser::GetSsrc(packet.cdata(), packet.data.size());
    RTC_CHECK(ssrc.has_value());
    media_type = ssrc_media_types_[*ssrc];
  }
  task_queue_.PostTask(
      [call = call_.get(), media_type, packet = std::move(packet)]() mutable {
        call->Receiver()->DeliverPacket(media_type, packet.data,
                                        packet.arrival_time.us());
      });
}

Encounted bugs

花了一周时间，代码已经移植到ns3。解决掉一些bug。
注释掉webrtc几句代码，这个判断导致数据不能发送。

//third_party/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
bool ModuleRtpRtcpImpl::TrySendPacket(RtpPacketToSend* packet,
                                      const PacedPacketInfo& pacing_info) {
  RTC_DCHECK(rtp_sender_);
  // TODO(sprang): Consider if we can remove this check.
  //if (!rtp_sender_->packet_generator.SendingMedia()) {
 //   return false;
 // }
  rtp_sender_->packet_sender.SendPacket(packet, pacing_info);
  return true;
}

另外就是ns3的packet对象不要再ns3之外的线程使用。这里为了减少内存的分配次数，Buffer被放进了g_freeList，非线程安全。

Buffer::Data *
Buffer::Create (uint32_t dataSize)
{
  NS_LOG_FUNCTION (dataSize);
  /* try to find a buffer correctly sized. */
  if (IS_UNINITIALIZED (g_freeList))
    {
      g_freeList = new Buffer::FreeList ();
    }
  else if (IS_INITIALIZED (g_freeList))
    {
      while (!g_freeList->empty ()) 
        {
          struct Buffer::Data *data = g_freeList->back ();
          g_freeList->pop_back ();
          if (data->m_size >= dataSize) 
            {
              data->m_count = 1;
              return data;
            }
          Buffer::Deallocate (data);
        }
    }
  struct Buffer::Data *data = Buffer::Allocate (dataSize);
  NS_ASSERT (data->m_count == 1);
  return data;
}

程序崩溃的一些提示：

corrupted double-linked list。
corrupted size vs precv_size

assert failed. cond="m_data->m_count == 1 || m_start == m_data->m_dirtyStart", file=../src/network/model/buffer.cc, line=321
terminate called without an active exception


0x00007fffef89619f in ns3::Buffer::Create (dataSize=1438)
    at ../src/network/model/buffer.cc:133
133	          if (data->m_size >= dataSize)

Results

Emulation mode (os time)

在这里插入图片描述

Simulation mode (simulator time)

You could clearly notice the difference. There are many working threads in emulation mode.
从图中结果可以看出，不同的GCC数据流不能收敛到相近的带宽。有同学可能会想，能否寻找机制，优化GCC算法的带宽公平性。这个估计是不能实现的。GCC采用的时延梯度作为拥塞信号。文献[1]指出，基于梯度的拥塞控制算法，均衡点不唯一。当时延梯度信号稳定的时候，算法进入均衡点。但是网络排队长度 $q (t)$ 有很多中变化情况，可以实现时延梯度 $\dot\theta$ 的稳定。

In particular, RTT-gradient approaches do not have a unique equilibrium point suggesting a lack of control over queue lengths. Intuitively, RTT-gradient approaches quickly adapt the sending rate to stabilize the RTT-gradient ( $\dot\theta=\frac{\dot q}{C}$ ) which in turn only stabilizes the queue length gradient $\dot q(t)$ but fails to control the absolute value of the queue length.