webrtc中的rtcp处理 | 易学教程

在webrtc中，处理rtcp，目前的版本是在, rtcp_sender.h 和 rtcp_sender.cpp中实现；

当收到rtcp packet时，调用过程，主要是以下三个函数：

接收-》解析-》响应；

IncomingPacket -》{ ParseCompoundPacket -》 TriggerCallbacksFromRtcpPacket }；

当然，还是实现了（根据rtcp）延时计算：

// Get rtt.
int32_t RTT(uint32_t remote_ssrc,
int64_t* last_rtt_ms,
int64_t* avg_rtt_ms,
int64_t* min_rtt_ms,
int64_t* max_rtt_ms) const;

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);

}

ParseCompoundPacket，是解析函数，直接看源码；

说明一下这个响应函数：

RTCPReceiver::TriggerCallbacksFromRtcpPacket；

这个函数，响应了，TMMBR and RTMB，NACK，和 Pli/Fir/Pli；

再说明一下 Pli/Fir/Pli：对关键帧的请求；

if (rtcp_intra_frame_observer_) {
RTC_DCHECK(!receiver_only_);
if ((packet_information.packet_type_flags & kRtcpPli) ||
(packet_information.packet_type_flags & kRtcpFir)) {
if (packet_information.packet_type_flags & kRtcpPli) {
RTC_LOG(LS_VERBOSE)
<< "Incoming PLI from SSRC " << packet_information.remote_ssrc;
} else {
RTC_LOG(LS_VERBOSE)
<< "Incoming FIR from SSRC " << packet_information.remote_ssrc;
}
rtcp_intra_frame_observer_->OnReceivedIntraFrameRequest(local_ssrc);
}
}

OnReceivedIntraFrameRequest函数会最后在本地产生一个相关的关键帧；

再看一个函数：

响应Fir，然后设置packet_type_flags；

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

for (const rtcp::Fir::Request& fir_request : fir.requests()) {
// Is it our sender that is requested to generate a new keyframe.
if (main_ssrc_ != fir_request.ssrc)
continue;

++packet_type_counter_.fir_packets;

int64_t now_ms = clock_->TimeInMilliseconds();
auto inserted = last_fir_.insert(std::make_pair(
fir.sender_ssrc(), LastFirStatus(now_ms, fir_request.seq_nr)));
if (!inserted.second) { // There was already an entry.
LastFirStatus* last_fir = &inserted.first->second;

// Check if we have reported this FIRSequenceNumber before.
if (fir_request.seq_nr == last_fir->sequence_number)
continue;

// Sanity: don't go crazy with the callbacks.
if (now_ms - last_fir->request_ms < kRtcpMinFrameLengthMs)
continue;

last_fir->request_ms = now_ms;
last_fir->sequence_number = fir_request.seq_nr;
}

// Received signal that we need to send a new key frame.

packet_information->packet_type_flags |= kRtcpFir;

}
}

来源：CSDN

作者：chinabinlang

链接：https://blog.csdn.net/chinabinlang/article/details/80132380

标签

rtcp

webrtc