webrtc-m79-音视频rtcp报文处理

1 RTCP 处理入口

PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
                                                 const uint8_t* packet,
                                                 size_t length) {
  TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
  // TODO(pbos): Make sure it's a valid packet.
  //             Return DELIVERY_UNKNOWN_SSRC if it can be determined that
  //             there's no receiver of the packet.
  if (received_bytes_per_second_counter_.HasSample()) {
    // First RTP packet has been received.
    received_bytes_per_second_counter_.Add(static_cast<int>(length));
    received_rtcp_bytes_per_second_counter_.Add(static_cast<int>(length));
  }
  bool rtcp_delivered = false;
  if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
    ReadLockScoped read_lock(*receive_crit_);
    for (VideoReceiveStream* stream : video_receive_streams_) { // video_receive_streams_ 中实际包含的是 webrtc::VideoReceiveStream
      if (stream->DeliverRtcp(packet, length))
        rtcp_delivered = true;
    }
  }
  if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
    ReadLockScoped read_lock(*receive_crit_);
    for (AudioReceiveStream* stream : audio_receive_streams_) { // audio_receive_streams_ 中实际包含的是 webrtc::AudioReceiveStream
      stream->DeliverRtcp(packet, length);
      rtcp_delivered = true;
    }
  }
  if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
    ReadLockScoped read_lock(*send_crit_);
    for (VideoSendStream* stream : video_send_streams_) {
      stream->DeliverRtcp(packet, length);
      rtcp_delivered = true;
    }
  }
  if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
    ReadLockScoped read_lock(*send_crit_);
    for (auto& kv : audio_send_ssrcs_) {
      kv.second->DeliverRtcp(packet, length);
      rtcp_delivered = true;
    }
  }

  if (rtcp_delivered) {
    event_log_->Log(std::make_unique<RtcEventRtcpPacketIncoming>(
        rtc::MakeArrayView(packet, length)));
  }

  return rtcp_delivered ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
}

2 视频RTCP 处理

bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
  return rtp_video_stream_receiver_.DeliverRtcp(packet, length); // RtpVideoStreamReceiver::DeliverRtcp
}

bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
                                         size_t rtcp_packet_length) {
  RTC_DCHECK_RUN_ON(&worker_task_checker_);

  if (!receiving_) {
    return false;
  }

  rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); // ModuleRtpRtcpImpl::IncomingRtcpPacket

  int64_t rtt = 0;
  rtp_rtcp_->RTT(config_.rtp.remote_ssrc, &rtt, nullptr, nullptr, nullptr);
  if (rtt == 0) {
    // Waiting for valid rtt.
    return true;
  }
  uint32_t ntp_secs = 0;
  uint32_t ntp_frac = 0;
  uint32_t rtp_timestamp = 0;
  uint32_t recieved_ntp_secs = 0;
  uint32_t recieved_ntp_frac = 0;
  if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, &recieved_ntp_secs, // ModuleRtpRtcpImpl::RemoteNTP
                           &recieved_ntp_frac, &rtp_timestamp) != 0) {
    // Waiting for RTCP.
    return true;
  }
  NtpTime recieved_ntp(recieved_ntp_secs, recieved_ntp_frac);
  int64_t time_since_recieved =
      clock_->CurrentNtpInMilliseconds() - recieved_ntp.ToMs();
  // Don't use old SRs to estimate time.
  if (time_since_recieved <= 1) {
    ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); // RemoteNtpTimeEstimator::UpdateRtcpTimestamp
  }

  return true;
}


void ModuleRtpRtcpImpl::IncomingRtcpPacket(const uint8_t* rtcp_packet,
                                           const size_t length) {
  rtcp_receiver_.IncomingPacket(rtcp_packet, length); // RTCPReceiver::IncomingPacket
}

void RTCPReceiver::IncomingPacket(const uint8_t* packet, size_t packet_size) {
  if (packet_size == 0) {
    RTC_LOG(LS_WARNING) << "Incoming empty RTCP packet";
    return;
  }

  PacketInformation packet_information;
  if (!ParseCompoundPacket(packet, packet + packet_size, &packet_information)) // 
    return;
  TriggerCallbacksFromRtcpPacket(packet_information);
}

3 音频RTCP 处理

void AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
  // TODO(solenberg): Tests call this function on a network thread, libjingle
  // calls on the worker thread. We should move towards always using a network
  // thread. Then this check can be enabled.
  // RTC_DCHECK(!thread_checker_.IsCurrent());
  channel_receive_->ReceivedRTCPPacket(packet, length); // ChannelReceive::ReceivedRTCPPacket
}


void ChannelReceive::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
  // Store playout timestamp for the received RTCP packet
  UpdatePlayoutTimestamp(true);

  // Deliver RTCP packet to RTP/RTCP module for parsing
  _rtpRtcpModule->IncomingRtcpPacket(data, length); // ModuleRtpRtcpImpl::IncomingRtcpPacket

  int64_t rtt = GetRTT();
  if (rtt == 0) {
    // Waiting for valid RTT.
    return;
  }

  uint32_t ntp_secs = 0;
  uint32_t ntp_frac = 0;
  uint32_t rtp_timestamp = 0;
  if (0 != _rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL, // ModuleRtpRtcpImpl::RemoteNTP
                                     &rtp_timestamp)) {
    // Waiting for RTCP.
    return;
  }

  {
    rtc::CritScope lock(&ts_stats_lock_);
    ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); // RemoteNtpTimeEstimator::UpdateRtcpTimestamp
  }
}


void RTCPReceiver::IncomingPacket(const uint8_t* packet, size_t packet_size) {
  if (packet_size == 0) {
    RTC_LOG(LS_WARNING) << "Incoming empty RTCP packet";
    return;
  }

  PacketInformation packet_information;
  if (!ParseCompoundPacket(packet, packet + packet_size, &packet_information)) // 
    return;
  TriggerCallbacksFromRtcpPacket(packet_information);
}

4 相关类图

webrtc::RtpVideoStreamReceiver 和 webrtc::voe::ChannelReceive 是相对应的关系，在二者的对象内都包含std::unique_ptr<RtpRtcp>、RemoteNtpTimeEstimator等，只不过一个用来处理视频，另外一个处理音频。