基于 Chromium M69版本

當(dāng) RTP 包完成拆分后镊掖，BaseChannel 把包到達(dá)時(shí)間抓換成微秒夹攒，然后通知 MediaChannel 處理收到的包仅颇。

// src/pc/channel.cc
void BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet) {
  // 重構(gòu)了 parsed_packet 里的 PacketTime
  OnPacketReceived(/*rtcp=*/false, parsed_packet.Buffer(), packet_time);
}
void BaseChannel::OnPacketReceived(bool rtcp,
                                   const rtc::CopyOnWriteBuffer& packet,
                                   const rtc::PacketTime& packet_time) {
  // 如果是第一個(gè) RTP 包則進(jìn)行額外處理
  invoker_.AsyncInvoke<void>(
      RTC_FROM_HERE, worker_thread_,
      Bind(&BaseChannel::ProcessPacket, this, rtcp, packet, packet_time));
}
void BaseChannel::ProcessPacket(bool rtcp,
                                const rtc::CopyOnWriteBuffer& packet,
                                const rtc::PacketTime& packet_time) {
  media_channel_->OnPacketReceived(&data, packet_time);
}

MediaChannel 是個(gè)父類受神，由 VoiceMediaChannel，VideoMediaChannel和DataMediaChannel實(shí)現(xiàn)魏颓，一下主要講視頻方面岭辣。WebRtcVideoChannel 繼承了 VideoMediaChannel，實(shí)現(xiàn)了 OnPacketReceived甸饱，響應(yīng) BaseChannel 的調(diào)用沦童。

WebRtcVideoChannel 通知 Call 遞交 RTP 包。如果遞交結(jié)果為DELIVERY_OK則說明包被正常遞交柜候，DELIVERY_PACKET_ERROR說明包無法正常解析搞动，DELIVERY_UNKNOWN_SSRC說明沒找到該包的 ssrc，最后返回渣刷。

// src/media/engine/webrtcvideoengine.cc
void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
                                          const rtc::PacketTime& packet_time) {
  const webrtc::PacketReceiver::DeliveryStatus delivery_result =
      call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
                                       webrtc_packet_time);
}

Call 判斷包是否能正常解析，是否能找到對(duì)應(yīng) ssrc矗烛，計(jì)算包到達(dá)時(shí)間辅柴，計(jì)算方式為 (timestamp_us + 500) / 1000，調(diào)用 RemoteEstimatorProxy 記錄每個(gè)包序列號(hào)對(duì)應(yīng)的到達(dá)時(shí)間用于 send-side congestion control瞭吃，然后把視頻包傳遞給 RtpStreamReceiverController 進(jìn)一步處理碌嘀。之后統(tǒng)計(jì)每秒收到的數(shù)據(jù)總大小和每秒收到的視頻數(shù)據(jù)總大小。

// src/call/call.cc
PacketReceiver::DeliveryStatus Call::DeliverPacket(
    MediaType media_type,
    rtc::CopyOnWriteBuffer packet,
    const PacketTime& packet_time) {
  // 如果是 RTCP 包則調(diào)用 DeliverRtcp 處理
  return DeliverRtp(media_type, std::move(packet), packet_time);
}
PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
                                                rtc::CopyOnWriteBuffer packet,
                                                const PacketTime& packet_time) {

  NotifyBweOfReceivedPacket(parsed_packet, media_type);
  video_receiver_controller_.OnRtpPacket(parsed_packet);
}

RtpStreamReceiverController 沒有其他操作直接傳給 RtpDemuxer 處理包歪架。

// src/call/rtp_stream_receiver_controller.cc
bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) {
  rtc::CritScope cs(&lock_);
  return demuxer_.OnRtpPacket(packet);
}

RtpDemuxer 根據(jù) MID股冗、RSID 或者 payload 類型獲取對(duì)應(yīng)的 RtpPacketSinkInterface。
視頻包由 RtpVideoStreamReceiver 繼續(xù)處理和蚪，其余還有 RtxReceiveStream止状、FlexfecReceiveStream、ChannelProxy攒霹。

// src/call/rtp_demuxer.cc
bool RtpDemuxer::OnRtpPacket(const RtpPacketReceived& packet) {
  RtpPacketSinkInterface* sink = ResolveSink(packet);
  if (sink != nullptr) {
    sink->OnRtpPacket(packet);
    return true;
  }
  return false;
}

RtpVideoStreamReceiver 負(fù)責(zé)處理常規(guī) RTP 包和由 FlexFEC 恢復(fù)的包怯疤。首先獲取包的 Header 信息，然后通知 RtpReceiver 正式接收 RTP 包催束，記錄非重傳包的接收數(shù)據(jù)集峦。最后調(diào)用二級(jí)sink繼續(xù)處理。

// src/video/rtp_video_stream_receiver.cc
void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
  packet.GetHeader(&header);
  ReceivePacket(packet.data(), packet.size(), header);
  rtp_receive_statistics_->IncomingPacket(header, packet.size(),
                                          IsPacketRetransmitted(header));
  secondary_sink->OnRtpPacket(packet);
}
void RtpVideoStreamReceiver::ReceivePacket(const uint8_t* packet,
                                           size_t packet_length,
                                           const RTPHeader& header) {
  if (header.payloadType == config_.rtp.red_payload_type) {
    ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
    return;
  }
  const uint8_t* payload = packet + header.headerLength;
  assert(packet_length >= header.headerLength);
  size_t payload_length = packet_length - header.headerLength;
  const auto pl =
      rtp_payload_registry_.PayloadTypeToPayload(header.payloadType);
  if (pl) {
    rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
                                     pl->typeSpecific);
  }
}

RtpReceiver 類由 RtpReceiverImpl 進(jìn)行實(shí)現(xiàn)，響應(yīng)上面對(duì) IncomingRtpPacket 的調(diào)用塔淤。RtpReceiver 負(fù)責(zé)處理 audio_level 信息摘昌，通知 RTPReceiverStrategy 解析 RTP 包，并區(qū)別遲到和重傳的包高蜂。

bool RtpReceiverImpl::IncomingRtpPacket(const RTPHeader& rtp_header,
                                        const uint8_t* payload,
                                        size_t payload_length,
                                        PayloadUnion payload_specific) {
  int32_t ret_val = rtp_media_receiver_->ParseRtpPacket(
      &webrtc_rtp_header, payload_specific, payload, payload_length,
      clock_->TimeInMilliseconds());
}

RTPReceiverStrategy 類由 RTPReceiverVideo 和 RTPReceiverAudio 進(jìn)行實(shí)現(xiàn)聪黎，視頻 RTP 包由 RTPReceiverVideo 負(fù)責(zé)具體解析 rtp_header 和 rtp_payload。

int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
                                         const PayloadUnion& specific_payload,
                                         const uint8_t* payload,
                                         size_t payload_length,
                                         int64_t timestamp_ms) {}