DSS 代码分析【SR包发送】_we only want to send an rtcp if the rtp packet was-CSDN博客

本文链接：https://blog.csdn.net/weixin_39799839/article/details/78604503

本文详细介绍了Darwin流媒体系统中RTPStream模块如何发送RTCP SR(发送端报告)。SR报告包含了发送方关于自身状态的重要信息，如发送的包数、字节数等，用于同步和质量监测。SR报告的发送频率取决于特定的间隔设置。

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Darwin 流媒体在发送RTP时会定期的发送SR发送端报告，具体调用过程如下

SR发送的函数具体实现如下：

void RTPStream::SendRTCPSR(const SInt64& inTime, Bool16 inAppendBye)
{
	// This will roll over, after which payloadByteCount will be all messed up.
	// But because it is a 32 bit number, that is bound to happen eventually,
	// and we are limited by the RTCP packet format in that respect, so this is
	// pretty much ok.
	UInt32 payloadByteCount = fByteCount - (12 * fPacketCount);

	RTCPSRPacket* theSR = fSession->GetSRPacket();
	theSR->SetSSRC(fSsrc);
	theSR->SetClientSSRC(fClientSSRC);
	//fLastNTPTimeStamp = fSession->GetNTPPlayTime() + OS::TimeMilli_To_Fixed64Secs(inTime - fSession->GetPlayTime());
	fLastNTPTimeStamp = OS::TimeMilli_To_1900Fixed64Secs(OS::Milliseconds()); //The time value should be filled in as late as possible.
	theSR->SetNTPTimestamp(fLastNTPTimeStamp);
	theSR->SetRTPTimestamp(fLastRTPTimestamp);
	theSR->SetPacketCount(fPacketCount);
	theSR->SetByteCount(payloadByteCount);
#if RTP_PACKET_RESENDER_DEBUGGING
	fResender.logprintf("Recommending ack timeout of: %d\n", fSession->GetBandwidthTracker()->RecommendedClientAckTimeout());
#endif
	theSR->SetAckTimeout(fSession->GetBandwidthTracker()->RecommendedClientAckTimeout());

	UInt32 thePacketLen = theSR->GetSRPacketLen();
	if (inAppendBye)
		thePacketLen = theSR->GetSRWithByePacketLen();

	QTSS_Error err = QTSS_NoErr;
	if (fTransportType == qtssRTPTransportTypeTCP)    // write out in interleave format on the RTSP TCP channel
	{
		UInt32  wasWritten;
		err = this->InterleavedWrite(theSR->GetSRPacket(), thePacketLen, &wasWritten, fRTCPChannel);
	}
	else
	{
		void *ptr = theSR->GetSRPacket();
		err = fSockets->GetSocketB()->SendTo(fRemoteAddr, fRemoteRTCPPort, ptr, thePacketLen);
		this->UDPMonitorWrite(ptr, thePacketLen, kIsRTCPPacket);
	}


	if (err == QTSS_NoErr)
		this->PrintPacketPrefEnabled((char *)theSR->GetSRPacket(), thePacketLen, (SInt32)RTPStream::rtcpSR); // if we are flow controlled this packet is not sent
}

SR包的详细格式请参照 http://blog.csdn.net/weixin_39799839/article/details/78586529

RTPStream::SendRTCPSR是在RTPStream::Write中被调用，当发送完RTP包后，会检查RTCP的发送周期是否到了，如果到了，就发送SR包

                        // Send an RTCP sender report if it's time. Again, we only want to send an
			// RTCP if the RTP packet was sent sucessfully
			// If doing rate-adaptation, then send an RTCP SR every seconds so that we get faster RTT feedback.
			UInt32 senderReportInterval = fStream3GPP->RateAdaptationEnabled() ? kSenderReportInterval3GPPInSecs : kSenderReportIntervalInSecs;
			if ((fSession->GetPlayFlags() & qtssPlayFlagsSendRTCP) &&
				(theTime > (fLastSenderReportTime + (senderReportInterval * 1000))))
			{
				fLastSenderReportTime = theTime;
				// CISCO comments
				// thePacket->packetTransmissionTime is
				// the expected transmission time, which
				// is what we should report in RTCP for
				// synchronization purposes, not theTime,
				// which is the actual transmission time.
				this->SendRTCPSR(thePacket->packetTransmitTime);
			}