gcc 算法核心

twcc 反馈获取延迟梯度
trendline (窗口，平滑系数，增益 => 最小二乘法计算斜率)，得出网络状态
网络状态 aimd 变更码率，得到估算当前码率

twcc 反馈路径

gcc 拥塞控制算法核心控制入口，想要实现自己的算法，继承NetworkControllerInterface类即可。

class NetworkControllerInterface {
 public:
  virtual ~NetworkControllerInterface() = default;
  // ...
  // Called when remotely calculated bitrate is received.
  ABSL_MUST_USE_RESULT virtual NetworkControlUpdate OnRemoteBitrateReport(
  // ...
  // Called when a protocol specific calculation of packet loss has been made.
  ABSL_MUST_USE_RESULT virtual NetworkControlUpdate OnTransportLossReport(
      TransportLossReport) = 0;
  // Called with per packet feedback regarding receive time.
  ABSL_MUST_USE_RESULT virtual NetworkControlUpdate OnTransportPacketsFeedback(
      TransportPacketsFeedback) = 0;
  // Called with network state estimate updates.
  ABSL_MUST_USE_RESULT virtual NetworkControlUpdate OnNetworkStateEstimate(
      NetworkStateEstimate) = 0;
};

webrtc 源代码中，pcc 和 goog_cc 都实现了该接口。下面 twcc 反馈的传导路径

void Port::OnReadPacket(const char* data, size_t size, const rtc::SocketAddress& addr, ProtocolType proto) {
void P2PTransportChannel::OnReadPacket(Connection* connection, const char* data, size_t len, int64_t packet_time_us) {
void DtlsTransport::OnReadPacket(rtc::PacketTransportInternal* transport, const char* data, size_t size, const int64_t& packet_time_us, int flags) {
bool DtlsTransport::HandleDtlsPacket(const char* data, size_t size) {
void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us)
PacketReceiver::DeliveryStatus Call::DeliverPacket( MediaType media_type, rtc::CopyOnWriteBuffer packet, int64_t packet_time_us)
bool RtpVideoStreamReceiver2::DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length) - 音频和视频都有对应的处理，这里指列举视频的处理
void ModuleRtpRtcpImpl2::IncomingRtcpPacket(const uint8_t* rtcp_packet, const size_t length) {
void RTCPReceiver::IncomingPacket(rtc::ArrayViewpacket)
void RTCPReceiver::TriggerCallbacksFromRtcpPacket
void RtpTransportControllerSend::OnTransportFeedback
NetworkControllerInterface::OnTransportPacketsFeedback

几点提示

Twcc 反馈的载体是 rtcp，所以自然处理的入口在 RTCPReceiver 里面
webrtc 存在一个较为复杂上下层协议栈交互的系统

PacketReceiver::DeliveryStatus Call::DeliverPacket(
    MediaType media_type,
    rtc::CopyOnWriteBuffer packet,
    int64_t packet_time_us) {
  if (IsRtcpPacket(packet)) {
    RTC_DCHECK_RUN_ON(network_thread_);
    DeliverRtcp(media_type, std::move(packet));
    return DELIVERY_OK;
  }

  RTC_DCHECK_RUN_ON(worker_thread_);
  return DeliverRtp(media_type, std::move(packet), packet_time_us);
}

其他

modules/remote_bitrate_estimator 基于 REMB 的旧版本，基本已经废弃了，其中几个类在新版本中也有使用 remb
modules/congestion_controller/receive_side_congestion_controller.cc remb 使用入口
modules/remote_bitrate_estimator remb 核心实现
modules/congestion_controller 基于 Transport Wide CC 的新版本，新旧版本中接收端的估算逻辑移到了发送端，其中的卡尔曼滤波算法也改成了线性回归算法
modules/congestion_controller/pcc Probe-based Congestion Control

25 June 2023 会持续更新

Dylan

webrtc-source-gcc

gcc 算法核心

twcc 反馈路径

其他