WebRTC源码之RTCPReceiver源码分析

WebRTC源码之RTCPReceiver源码分析

---

WebRTC专题开嗨鸭 ！！！

一、 WebRTC 线程模型

1、WebRTC中线程模型和常见线程模型介绍

2、WebRTC网络PhysicalSocketServer之WSAEventselect模型使用

二、 WebRTC媒体协商

1、WebRTC媒体协商之SDP中JsepSessionDescription类结构分析

2、WebRTC媒体协商之CreatePeerConnectionFactory、CreatePeerConnection、CreateOffer

3、WebRTC之证书(certificate)生成的时机分析

4、WebRTC源码之RtpTransceiver添加视频轨道的AddTrack函数中桥接模式的流程分析

三、 WebRTC 音频数据采集

1、WebRTC源码之音频设备播放流程源码分析

2、WebRTC源码之音频设备的录制流程源码分析

四、 WebRTC 音频引擎(编解码和3A算法)

五、 WebRTC 视频数据采集

六、 WebRTC 视频引擎( 编解码)

七、 WebRTC 网络传输

1、WebRTC的ICE之STUN协议

2、WebRTC的ICE之Dtls/SSL/TLSv1.x协议详解

八、 WebRTC服务质量(Qos)

1、WebRTC中RTCP协议详解

2、WebRTC中RTP协议详解

3、WebRTC之NACK、RTX 在什么时机判断丢包发送NACK请求和RTX丢包重传

4、WebRTC源码之视频质量统计数据的数据结构分析

5、WebRTC源码之RTCPReceiver源码分析

九、 NetEQ

十、 Simulcast与SVC

前言

WebRTC是音视频行业的标杆， 如果要学习音视频， WebRTC是进入音视频行业最好方法， 里面可以有成熟方案， 例如：音频中3A 算法、网络评估、自适应码流、Simulcast、SVC等等 ， 非常适合刚刚进入音视频行业小伙伴哈^_ 我也是哦， 以后再音视频行业长期打算的小伙伴的学习项目。 里面有大量知识点

---

提示：以下是本篇文章正文内容，下面案例可供参考

一、 RTCP接受数据的流程的堆栈信息的

1、网络io 线程读取数据

AllocationSequence::OnReadPacket(rtc::AsyncPacketSocket* socket,  const char* data, size_t size, const rtc::SocketAddress& remote_addr, const int64_t& packet_time_us)
UDPPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket, const char* data, size_t size, const rtc::SocketAddress& remote_addr, int64_t packet_time_us)
UDPPort::OnReadPacket(rtc::AsyncPacketSocket* socket, const char* data, size_t size, const rtc::SocketAddress& remote_addr, const int64_t& packet_time_us)
onnection::OnReadPacket(const char* data, size_t size, int64_t packet_time_us)
P2PTransportChannel::OnReadPacket(Connection* connection, const char* data, size_t len, int64_t packet_time_us)
DtlsTransport::OnReadPacket(rtc::PacketTransportInternal* transport, const char* data, size_t size, const int64_t& packet_time_us, int flags)
RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport, const char* data, size_t len, const int64_t& packet_time_us, int flags)
SrtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us)
RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer packet, int64_t packet_time_us)										
RtpDemuxer::OnRtpPacket(const RtpPacketReceived& packet)
BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet)
BaseChannel::OnPacketReceived(bool rtcp, const rtc::CopyOnWriteBuffer& packet, int64_t packet_time_us) 

2、 线程切换的代码

invoker_.AsyncInvoke<void>(
      RTC_FROM_HERE, worker_thread_,
      Bind(&BaseChannel::ProcessPacket, this, rtcp, packet, packet_time_us));
]

// received 

BaseChannel::ProcessPacket(bool rtcp, const rtc::CopyOnWriteBuffer& packet, int64_t packet_time_us)
WebRtcVideoChannel::OnRtcpReceived
Call::DeliverPacket
Call::DeliverRtcp
VideoSendStream::DeliverRtcp 
VideoSendStreamImpl::DeliverRtcp 
RtpVideoSender::DeliverRtcp 
ModuleRtpRtcpImpl::IncomingRtcpPacket(const uint8_t* rtcp_packet, const size_t length)
RTCPReceiver::IncomingPacket(const uint8_t* packet, size_t packet_size)  --> RTCPReceiver模块接受数据并读取数据格式
RTCPReceiver::TriggerCallbacksFromRtcpPacket(const PacketInformation& packet_information)
RtpTransportControllerSend::OnTransportFeedback(const rtcp::TransportFeedback& feedback)

3、 线程切换 gcc

GoogCcNetworkController::OnTransportPacketsFeedback(TransportPacketsFeedback report)

二、 RTCPReceiver::IncomingPacket方法读取RTCP数据的格式

1、 ParseCompoundPacket解析rtcp的头数据

void RTCPReceiver::IncomingPacket(const uint8_t* packet, size_t packet_size) {
  if (packet_size == 0) {
    RTC_LOG(LS_WARNING) << "Incoming empty RTCP packet";
    return;
  }
  // TODO@chensong 20220909 根据对端反馈信息处理
  // TODO@chensong 2022-10-19   解析RTCP 数据包的格式 
  PacketInformation packet_information;
  if (!ParseCompoundPacket(packet, packet + packet_size, &packet_information)) {
    return;
  }
  TriggerCallbacksFromRtcpPacket(packet_information);
}


CommonHeader rtcp_block;
  for (const uint8_t* next_block = packet_begin; next_block != packet_end; next_block = rtcp_block.NextPacket()) 
  {
    ptrdiff_t remaining_blocks_size = packet_end - next_block;
    RTC_DCHECK_GT(remaining_blocks_size, 0);
    //rtcp包有统一头格式读取方法
    if (!rtcp_block.Parse(next_block, remaining_blocks_size)) 
	{
      if (next_block == packet_begin)
	  {
        // Failed to parse 1st header, nothing was extracted from this packet.
        RTC_LOG(LS_WARNING) << "Incoming invalid RTCP packet";
        return false;
      }
      ++num_skipped_packets_;
      break;
    }
***
}

2、 rtcp包有统一头格式读取方法

//  webrtc\src\modules\rtp_rtcp\source\rtcp_packet\common_header.cc
//    0                   1           1       2                   3
//    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// 0 |V=2|P|   C/F   |
//   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// 1                 |  Packet Type  |
//   ----------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// 2                                 |             length            |
//   --------------------------------+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// TODO@chensong 2022-12-25 
// Common header for all RTCP packets, 4 octets.
bool CommonHeader::Parse(const uint8_t* buffer, size_t size_bytes) 
{
  const uint8_t kVersion = 2;

  if (size_bytes < kHeaderSizeBytes) 
  {
    RTC_LOG(LS_WARNING)
        << "Too little data (" << size_bytes << " byte"
        << (size_bytes != 1 ? "s" : "")
        << ") remaining in buffer to parse RTCP header (4 bytes).";
    return false;
  }
  // rtcp 版本
  uint8_t version = buffer[0] >> 6;
  if (version != kVersion) 
  {
    RTC_LOG(LS_WARNING) << "Invalid RTCP header: Version must be "
                        << static_cast<int>(kVersion) << " but was "
                        << static_cast<int>(version);
    return false;
  }

  // 是否有扩展的数据包
  bool has_padding = (buffer[0] & 0x20) != 0;
  count_or_format_ = buffer[0] & 0x1F;
  // rtcp 包类型
  packet_type_ = buffer[1];

  // rtcp 包中数据大小 读取4个字节 就是32bit
  payload_size_ = ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) * 4;

  // rtcp 包中实际数据开始地址的位置
  payload_ = buffer + kHeaderSizeBytes /*default kHeaderSizeBytes = 4*/;
  padding_size_ = 0;

  if (size_bytes < kHeaderSizeBytes + payload_size_) 
  {
    RTC_LOG(LS_WARNING) << "Buffer too small (" << size_bytes
                        << " bytes) to fit an RtcpPacket with a header and "
                        << payload_size_ << " bytes.";
    return false;
  }

  if (has_padding) 
  {
    if (payload_size_ == 0) 
	{
      RTC_LOG(LS_WARNING)
          << "Invalid RTCP header: Padding bit set but 0 payload "
             "size specified.";
      return false;
    }

    padding_size_ = payload_[payload_size_ - 1];
    if (padding_size_ == 0) 
	{
      RTC_LOG(LS_WARNING)
          << "Invalid RTCP header: Padding bit set but 0 padding "
             "size specified.";
      return false;
    }
    if (padding_size_ > payload_size_) 
	{
      RTC_LOG(LS_WARNING) << "Invalid RTCP header: Too many padding bytes ("
                          << padding_size_ << ") for a packet payload size of "
                          << payload_size_ << " bytes.";
      return false;
    }
    payload_size_ -= padding_size_;
  }
  return true;
}

然后根据packet_type的不同类型进行处理

三、 rtcp::SenderReport::kPacketType（200）

1、SR数据包格式

    Sender report (SR) (RFC 3550).
     0                   1                   2                   3
     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |V=2|P|    RC   |   PT=SR=200   |             length            |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  0 |                         SSRC of sender                        |
    +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
  4 |              NTP timestamp, most significant word             |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
  8 |             NTP timestamp, least significant word             |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 12 |                         RTP timestamp                         |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 16 |                     sender's packet count                     |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 20 |                      sender's octet count                     |
 24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

2、 SR的数据解析

bool SenderReport::Parse(const CommonHeader& packet) 
{
  RTC_DCHECK_EQ(packet.type(), kPacketType);

  const uint8_t report_block_count = packet.count();
  if (packet.payload_size_bytes() < kSenderBaseLength/*24*/ + report_block_count * ReportBlock::kLength /*24*/) 
  {
    RTC_LOG(LS_WARNING) << "Packet is too small to contain all the data.";
    return false;
  }
  // Read SenderReport header.
  const uint8_t* const payload = packet.payload();
  // 发送端ssrc
  sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&payload[0]);
  uint32_t secs = ByteReader<uint32_t>::ReadBigEndian(&payload[4]);
  uint32_t frac = ByteReader<uint32_t>::ReadBigEndian(&payload[8]);
  ntp_.Set(secs, frac);
  // rtp 网络时间戳
  rtp_timestamp_ = ByteReader<uint32_t>::ReadBigEndian(&payload[12]);
  // 发送的总包数
  sender_packet_count_ = ByteReader<uint32_t>::ReadBigEndian(&payload[16]);
  // 总共发送数据包量
  sender_octet_count_ = ByteReader<uint32_t>::ReadBigEndian(&payload[20]);
  report_blocks_.resize(report_block_count);
  const uint8_t* next_block = payload + kSenderBaseLength;
  for (ReportBlock& block : report_blocks_) 
  {
    bool block_parsed = block.Parse(next_block, ReportBlock::kLength);
    RTC_DCHECK(block_parsed);
    next_block += ReportBlock::kLength;
  }
  // Double check we didn't read beyond provided buffer.
  RTC_DCHECK_LE(next_block - payload, static_cast<ptrdiff_t>(packet.payload_size_bytes()));
  return true;
}

3、 RTCP report block 反馈包格式

From RFC 3550, RTP: A Transport Protocol for Real-Time Applications.

RTCP report block (RFC 3550).

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
 0 |                 SSRC_1 (SSRC of first source)                 |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 4 | fraction lost |       cumulative number of packets lost       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 8 |           extended highest sequence number received           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
12 |                      interarrival jitter                      |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
16 |                         last SR (LSR)                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
20 |                   delay since last SR (DLSR)                  |
24 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+

4、RTCP report block 反馈包格式解析

bool ReportBlock::Parse(const uint8_t* buffer, size_t length) 
{
  RTC_DCHECK(buffer != nullptr);
  if (length < ReportBlock::kLength)
  {
    RTC_LOG(LS_ERROR) << "Report Block should be 24 bytes long";
    return false;
  }
  // 接收到的媒体源ssrc
  source_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]);
  // TODO@chensong 2022-10-19  丢包率 fraction_lost
  fraction_lost_ = buffer[4];
  // 接收开始丢包总数， 迟到包不算丢包，重传有可以导致负数
  cumulative_lost_ = ByteReader<int32_t, 3>::ReadBigEndian(&buffer[5]);
  // 低16位表示收到的最大seq，高16位表示seq循环次数
  extended_high_seq_num_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
  // rtp包到达时间间隔的统计方差
  jitter_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[12]);
  // ntp时间戳的中间32位
  last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[16]);
  // 记录上一个接收SR的时间与上一个发送SR的时间差
  delay_since_last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[20]);

  return true;
}

5、 SR包信息保持处理

大致是包RTCP report block包数据进行再加工 重要处理rtt, 然后把 RTCP report blocb包全部数据包抛到gcc模块中去

void RTCPReceiver::HandleSenderReport(const CommonHeader& rtcp_block, PacketInformation* packet_information) 
{
  rtcp::SenderReport sender_report;
  if (!sender_report.Parse(rtcp_block)) 
  {
    ++num_skipped_packets_;
    return;
  }

  const uint32_t remote_ssrc = sender_report.sender_ssrc();

  packet_information->remote_ssrc = remote_ssrc;

  UpdateTmmbrRemoteIsAlive(remote_ssrc);

  // Have I received RTP packets from this party?
  if (remote_ssrc_ == remote_ssrc) 
  {
    // Only signal that we have received a SR when we accept one.
    packet_information->packet_type_flags |= kRtcpSr;

    remote_sender_ntp_time_ = sender_report.ntp();
    remote_sender_rtp_time_ = sender_report.rtp_timestamp();
    last_received_sr_ntp_ = TimeMicrosToNtp(clock_->TimeInMicroseconds());
  } 
  else 
  {
    // We will only store the send report from one source, but
    // we will store all the receive blocks.
    packet_information->packet_type_flags |= kRtcpRr;
  }

  for (const rtcp::ReportBlock& report_block : sender_report.report_blocks()) 
  {
    HandleReportBlock(report_block, packet_information, remote_ssrc);
  }
}

void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block, PacketInformation* packet_information, uint32_t remote_ssrc) 
{
  // This will be called once per report block in the RTCP packet.
  // We filter out all report blocks that are not for us.
  // Each packet has max 31 RR blocks.
  //
  // We can calc RTT if we send a send report and get a report block back.

  // |report_block.source_ssrc()| is the SSRC identifier of the source to
  // which the information in this reception report block pertains.

  // Filter out all report blocks that are not for us.
  // TODO@chensong 2022-12-26 
  //这将在RTCP数据包中的每个报告块调用一次。 
  //我们过滤掉所有不适合我们的报告块。 
  //每个数据包最多有31个RR块。 
  //如果我们发送发送报告并得到报告块，我们可以计算RTT。 
  //|report_block.source_ssrc（）|是源的ssrc标识符 
  //该接收报告块中的信息与之相关。 
  //过滤掉所有不适合我们的报告块。
  if (registered_ssrcs_.count(report_block.source_ssrc()) == 0) 
  {
    return;
  }

  last_received_rb_ms_ = clock_->TimeInMilliseconds();
  // TODO@chensong 2022-12-26  没有该ssrc在received_report_blocks_中map中正好插入
  ReportBlockWithRtt* report_block_info = &received_report_blocks_[report_block.source_ssrc()][remote_ssrc];
  report_block_info->report_block.sender_ssrc = remote_ssrc;
  report_block_info->report_block.source_ssrc = report_block.source_ssrc();
  report_block_info->report_block.fraction_lost = report_block.fraction_lost();
  report_block_info->report_block.packets_lost = report_block.cumulative_lost_signed();
  if (report_block.extended_high_seq_num() > report_block_info->report_block.extended_highest_sequence_number) 
  {
    // We have successfully delivered new RTP packets to the remote side after
    // the last RR was sent from the remote side.
    last_increased_sequence_number_ms_ = clock_->TimeInMilliseconds();
  }
  report_block_info->report_block.extended_highest_sequence_number = report_block.extended_high_seq_num();
  report_block_info->report_block.jitter = report_block.jitter();
  report_block_info->report_block.delay_since_last_sender_report = report_block.delay_since_last_sr();
  report_block_info->report_block.last_sender_report_timestamp = report_block.last_sr();

  int64_t rtt_ms = 0;
  uint32_t send_time_ntp = report_block.last_sr();
  // RFC3550, section 6.4.1, LSR field discription states:
  // If no SR has been received yet, the field is set to zero.
  // Receiver rtp_rtcp module is not expected to calculate rtt using
  // Sender Reports even if it accidentally can.

  // TODO(nisse): Use this way to determine the RTT only when |receiver_only_|
  // is false. However, that currently breaks the tests of the
  // googCaptureStartNtpTimeMs stat for audio receive streams. To fix, either
  // delete all dependencies on RTT measurements for audio receive streams, or
  // ensure that audio receive streams that need RTT and stats that depend on it
  // are configured with an associated audio send stream.
  if (send_time_ntp != 0) 
  {
    uint32_t delay_ntp = report_block.delay_since_last_sr();
    // Local NTP time.
	// 微妙 
    uint32_t receive_time_ntp = CompactNtp(TimeMicrosToNtp(clock_->TimeInMicroseconds()));

    // RTT in 1/(2^16) seconds.
    uint32_t rtt_ntp = receive_time_ntp - delay_ntp /*发送时间与接收到时间差值*/ - send_time_ntp;
    // Convert to 1/1000 seconds (milliseconds).
	// 微妙转换 毫秒级
    rtt_ms = CompactNtpRttToMs(rtt_ntp);
    if (rtt_ms > report_block_info->max_rtt_ms) 
	{
      report_block_info->max_rtt_ms = rtt_ms;
    }

    if (report_block_info->num_rtts == 0 || rtt_ms < report_block_info->min_rtt_ms)
	{
      report_block_info->min_rtt_ms = rtt_ms;
    }

    report_block_info->last_rtt_ms = rtt_ms;
    report_block_info->sum_rtt_ms += rtt_ms;
    ++report_block_info->num_rtts;

    packet_information->rtt_ms = rtt_ms;
  }

  packet_information->report_blocks.push_back(report_block_info->report_block);
}

四、rtcp::ReceiverReport::kPacketType （201）

1、 RTCP receiver report包格式

RTCP receiver report (RFC 3550).

  0                   1                   2                   3
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |V=2|P|    RC   |   PT=RR=201   |             length            |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |                     SSRC of packet sender                     |
 +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
 |                         report block(s)                       |
 |                            ....                               |

2、读取数据包格式

bool ReceiverReport::Parse(const CommonHeader& packet) 
{
  RTC_DCHECK_EQ(packet.type(), kPacketType);

  const uint8_t report_blocks_count = packet.count();

  if (packet.payload_size_bytes() < kRrBaseLength + report_blocks_count * ReportBlock::kLength) 
  {
    RTC_LOG(LS_WARNING) << "Packet is too small to contain all the data.";
    return false;
  }
  // 发送者ssrc
  sender_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(packet.payload());

  const uint8_t* next_report_block = packet.payload() + kRrBaseLength;

  report_blocks_.resize(report_blocks_count);
  for (ReportBlock& block : report_blocks_)
  {
    block.Parse(next_report_block, ReportBlock::kLength);
    next_report_block += ReportBlock::kLength;
  }

  RTC_DCHECK_LE(next_report_block - packet.payload(),
                static_cast<ptrdiff_t>(packet.payload_size_bytes()));
  return true;
}

3、 RR数据保持与SR包保持一样然后抛到GCC模块中去了

void RTCPReceiver::HandleReceiverReport(const CommonHeader& rtcp_block, PacketInformation* packet_information) 
{
  rtcp::ReceiverReport receiver_report;
  if (!receiver_report.Parse(rtcp_block)) 
  {
    ++num_skipped_packets_;
    return;
  }

  const uint32_t remote_ssrc = receiver_report.sender_ssrc();

  packet_information->remote_ssrc = remote_ssrc;

  UpdateTmmbrRemoteIsAlive(remote_ssrc);

  packet_information->packet_type_flags |= kRtcpRr;

  for (const ReportBlock& report_block : receiver_report.report_blocks()) 
  {
	  // TODO@chensong 2022-12-26 和SR包反馈包的处理流程一样的然后抛到GCC模块中去了
    HandleReportBlock(report_block, packet_information, remote_ssrc);
  }
}

五、rtcp::Rtpfb::kPacketType （205）WebRTC中扩展字段

1、rtcp::Nack::kFeedbackMessageType（1）丢包重传

① nack格式

 RFC 4585: Feedback format.

 Common packet format:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|   FMT   |       PT      |          length               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 0 |                  SSRC of packet sender                        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 4 |                  SSRC of media source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :            Feedback Control Information (FCI)                 :
   :                                                               :

 Generic NACK (RFC 4585).

 FCI:
    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |            PID                |             BLP               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

看上面主要的Generic NACK格式是丢包数据（32bit）

1. 前面16个bit是first_seq的序号
2. 后面的16个bit是 索引（index），如果对应索引bit是1就是丢包了， 对应丢包的seq 序号 = （first_seq + 1 + index）

② nack的子包解析

void Nack::Unpack()
{
  RTC_DCHECK(packet_ids_.empty());
  RTC_DCHECK(!packed_.empty());
  for (const PackedNack& item : packed_)
  {
    packet_ids_.push_back(item.first_pid);
    uint16_t pid = item.first_pid + 1;
    // first_pid :  是记录第一个包seq的
    // bitmask   :  是掩码 1bit  是0 是没有丢包， 是1是丢包了
    for (uint16_t bitmask = item.bitmask; bitmask != 0; bitmask >>= 1, ++pid)
    {
      if (bitmask & 1)
      {
          packet_ids_.push_back(pid);
      }
    }
  }
}
bool Nack::Parse(const CommonHeader& packet)
{
  RTC_DCHECK_EQ(packet.type(), kPacketType);
  RTC_DCHECK_EQ(packet.fmt(), kFeedbackMessageType);

  if (packet.payload_size_bytes() < kCommonFeedbackLength + kNackItemLength) {
    RTC_LOG(LS_WARNING) << "Payload length " << packet.payload_size_bytes()
                        << " is too small for a Nack.";
    return false;
  }
  // nack 结点的个数  nack数据的大小是固定的  --> nack item 一个 4bit
  size_t nack_items = (packet.payload_size_bytes() - kCommonFeedbackLength /*8*/) / kNackItemLength /*4*/;

  // 读取sender ssrc 4个bit
  //     media ssrc 4个bit
  ParseCommonFeedback(packet.payload());
  const uint8_t* next_nack = packet.payload() + kCommonFeedbackLength;

  packet_ids_.clear();
  packed_.resize(nack_items);
  for (size_t index = 0; index < nack_items; ++index)
  {
    packed_[index].first_pid = ByteReader<uint16_t>::ReadBigEndian(next_nack);
    packed_[index].bitmask = ByteReader<uint16_t>::ReadBigEndian(next_nack + 2);
    next_nack += kNackItemLength;
  }
  // 解码nack的数据包   ->丢包的数据放到packet_ids中去
  Unpack();

  return true;
}

③ nack包的handler 数据包

void RTCPReceiver::HandleNack(const CommonHeader& rtcp_block,
                              PacketInformation* packet_information) {
  rtcp::Nack nack;
  if (!nack.Parse(rtcp_block))
  {
    ++num_skipped_packets_;
    return;
  }

  if (receiver_only_ || main_ssrc_ != nack.media_ssrc())  // Not to us.
  {
    return;
  }
  // 把丢包的seq插入nack_sequence_numbers中去请求重新发送seq包
  packet_information->nack_sequence_numbers.insert(packet_information->nack_sequence_numbers.end(), nack.packet_ids().begin(), nack.packet_ids().end());
    
    // 把丢包的seq的序号放到数据统计中去
  for (uint16_t packet_id : nack.packet_ids())
  {
    nack_stats_.ReportRequest(packet_id);
  }

  if (!nack.packet_ids().empty())
  {
    packet_information->packet_type_flags |= kRtcpNack;
      // 记录nack丢包请求的次数
    ++packet_type_counter_.nack_packets;
      // 记录总共nack包数量
    packet_type_counter_.nack_requests = nack_stats_.requests();
      // 超时的包
    packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
  }
}

④ nack的seq序号重新发送

// TODO@chensong 发送RTX丢包信息
  if (!receiver_only_ && (packet_information.packet_type_flags & kRtcpNack)) 
  {
    if (!packet_information.nack_sequence_numbers.empty()) 
	{
      RTC_LOG(LS_VERBOSE) << "Incoming NACK length: " << packet_information.nack_sequence_numbers.size();
      // 请求重新发送seq的包   ModuleRtpRtcpImpl->OnReceivedNack
      rtp_rtcp_->OnReceivedNack(packet_information.nack_sequence_numbers);
    }
  }

rtp_rtcp_impl.cc

void ModuleRtpRtcpImpl::OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers) 
{
  if (!rtp_sender_)
  {
    return;
  }
  // TODO@chensong 2022-12-20 发送包掉包的数据统计模块
  for (uint16_t nack_sequence_number : nack_sequence_numbers)
  {
    send_loss_stats_.AddLostPacket(nack_sequence_number);
  }
  // TODO@chensong 2022-12-20  判断是否符合重新发送seq的数据包
  if (!rtp_sender_->StorePackets() || nack_sequence_numbers.size() == 0) 
  {
    return;
  }
  // Use RTT from RtcpRttStats class if provided.
  int64_t rtt = rtt_ms();
  if (rtt == 0) 
  {
    rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
  }
  // TODO@chensong 2022-12-20 重新发送包插入到发送队列中去  RTPSender->OnReceivedNack
  rtp_sender_->OnReceivedNack(nack_sequence_numbers, rtt);
}

rtp_sender

int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
  // Try to find packet in RTP packet history. Also verify RTT here, so that we
  // don't retransmit too often.
  absl::optional<RtpPacketHistory::PacketState> stored_packet = packet_history_.GetPacketState(packet_id);
  if (!stored_packet) 
  {
    // Packet not found.
    return 0;
  }

  const int32_t packet_size = static_cast<int32_t>(stored_packet->packet_size);

  // Skip retransmission rate check if not configured.
  if (retransmission_rate_limiter_) 
  {
    // Check if we're overusing retransmission bitrate.
    // TODO(sprang): Add histograms for nack success or failure reasons.
    if (!retransmission_rate_limiter_->TryUseRate(packet_size)) 
	{
      return -1;
    }
  }
  // TODO@chensong 2022-12-20 发送seq sum包 插入到发送队列中去
  if (paced_sender_) 
  {
    // Convert from TickTime to Clock since capture_time_ms is based on
    // TickTime.
      //  把要发送包seq序号队列中去插入发送队列中去
    int64_t corrected_capture_tims_ms = stored_packet->capture_time_ms + clock_delta_ms_;
    paced_sender_->InsertPacket(RtpPacketSender::kNormalPriority, stored_packet->ssrc, stored_packet->rtp_sequence_number, corrected_capture_tims_ms, stored_packet->packet_size, true);

    return packet_size;
  }

  std::unique_ptr<RtpPacketToSend> packet = packet_history_.GetPacketAndSetSendTime(packet_id);
  if (!packet) {
    // Packet could theoretically time out between the first check and this one.
    return 0;
  }

  const bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
  if (!PrepareAndSendPacket(std::move(packet), rtx, true, PacedPacketInfo()))
  {
    return -1;
  }

  return packet_size;
}


void RTPSender::OnReceivedNack( const std::vector<uint16_t>& nack_sequence_numbers, int64_t avg_rtt) 
{
    // 这边为什么要平均增加+5呢 ？？？？
  packet_history_.SetRtt(5 + avg_rtt);
  for (uint16_t seq_no : nack_sequence_numbers) 
  {
    const int32_t bytes_sent = ReSendPacket(seq_no);
    if (bytes_sent < 0) 
	{
      // Failed to send one Sequence number. Give up the rest in this nack.
      RTC_LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
                          << ", Discard rest of packets.";
      break;
    }
  }
}

总结

WebRTC源码分析地址：https://github.com/chensongpoixs/cwebrtc