webrtc 中 Rtcp NACK 接收处理流程
webrtc_m74
1:
解析 rtcp ，如果是 NACK，那么解析；主要是解析相关的类型，和 获取相关参数；
case rtcp::Rtpfb::kPacketType:
switch (rtcp_block.fmt()) {
case rtcp::Nack::kFeedbackMessageType:
HandleNack(rtcp_block, packet_information);
2：
上面解析 rtcp 后，如果 rtcp 格式有效，那么任何rtcp都会触发相关的函数；
这个函数，封装了相关 rtcp 的处理回调函数；
TriggerCallbacksFromRtcpPacket(packet_information);
3:
webrtc 处理 Nack ，主要就是重发 没收到的 rtp 包，从历史记录中查找，发送；
void RTPSender::OnReceivedNack(
const std::vector<uint16_t>& nack_sequence_numbers,
int64_t avg_rtt) {
packet_history_.SetRtt(5 + avg_rtt);
for (uint16_t seq_no : nack_sequence_numbers) {
const int32_t bytes_sent = ReSendPacket(seq_no); //再次发送，这函数名称很好；
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;
}
}
}
4：
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.
//上面注释很好了，从 packet_history_ 找到 Nack 中的 packet ;
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;
}
}
if (paced_sender_) { //一般都有；
// Convert from TickTime to Clock since capture_time_ms is based on
// TickTime.
int64_t corrected_capture_tims_ms =
stored_packet->capture_time_ms + clock_delta_ms_;
//添加到 paced_sender_ 中，pacer 中 有一定的排序规则，一般 Nack 的 packet 会优先发送；
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; 完成，返回；
}
// 如果没有 paced_sender_ ，最后直接 RTPSender::SendPacketToNetwork；
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;
// RTPSender::SendPacketToNetwork；
if (!PrepareAndSendPacket(std::move(packet), rtx, true, PacedPacketInfo()))
return -1;
return packet_size;
}