live555 源码分析：播放启动

本文分析 live555 中，流媒体播放启动，数据开始通过 RTP/RTCP 传输的过程。

如我们在 live555 源码分析：子会话 SETUP 中看到的，一个流媒体子会话的播放启动，由 StreamState::startPlaying 完成：

void OnDemandServerMediaSubsession::startStream(unsigned clientSessionId,    void* streamToken,    TaskFunc* rtcpRRHandler,    void* rtcpRRHandlerClientData,    unsigned short& rtpSeqNum,    unsigned& rtpTimestamp,    ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,    void* serverRequestAlternativeByteHandlerClientData) {  StreamState* streamState = (StreamState*)streamToken;  Destinations* destinations    = (Destinations*)(fDestinationsHashTable->Lookup((char const*)clientSessionId));  if (streamState != NULL) {    streamState->startPlaying(destinations, clientSessionId,        rtcpRRHandler, rtcpRRHandlerClientData,        serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);    RTPSink* rtpSink = streamState->rtpSink(); // alias    if (rtpSink != NULL) {      rtpSeqNum = rtpSink->currentSeqNo();      rtpTimestamp = rtpSink->presetNextTimestamp();    }  }}

在这个函数中，首先找到子会话的目标地址，也就是客户端的 IP 地址，和用于接收 RTP/RTCP 的端口号，然后通过 StreamState::startPlaying() 启动播放，最后将 RTP 包的初始序列号和初始时间戳返回给调用者，也就是 RTSPServer，并由后者返回给客户端，以用于客户端的播放同步。

StreamState::startPlaying() 的实现是这样的：

void StreamState::startPlaying(Destinations* dests, unsigned clientSessionId,    TaskFunc* rtcpRRHandler, void* rtcpRRHandlerClientData,    ServerRequestAlternativeByteHandler* serverRequestAlternativeByteHandler,    void* serverRequestAlternativeByteHandlerClientData) {  if (dests == NULL) return;  if (fRTCPInstance == NULL && fRTPSink != NULL) {    // Create (and start) a 'RTCP instance' for this RTP sink:    fRTCPInstance = fMaster.createRTCP(fRTCPgs, fTotalBW, (unsigned char*)fMaster.fCNAME, fRTPSink);        // Note: This starts RTCP running automatically    fRTCPInstance->setAppHandler(fMaster.fAppHandlerTask, fMaster.fAppHandlerClientData);  }  if (dests->isTCP) {    // Change RTP and RTCP to use the TCP socket instead of UDP:    if (fRTPSink != NULL) {      fRTPSink->addStreamSocket(dests->tcpSocketNum, dests->rtpChannelId);      RTPInterface::setServerRequestAlternativeByteHandler(fRTPSink->envir(), dests->tcpSocketNum,        serverRequestAlternativeByteHandler, serverRequestAlternativeByteHandlerClientData);        // So that we continue to handle RTSP commands from the client    }    if (fRTCPInstance != NULL) {      fRTCPInstance->addStreamSocket(dests->tcpSocketNum, dests->rtcpChannelId);      fRTCPInstance->setSpecificRRHandler(dests->tcpSocketNum, dests->rtcpChannelId,          rtcpRRHandler, rtcpRRHandlerClientData);    }  } else {    // Tell the RTP and RTCP 'groupsocks' about this destination    // (in case they don't already have it):    if (fRTPgs != NULL) fRTPgs->addDestination(dests->addr, dests->rtpPort, clientSessionId);    if (fRTCPgs != NULL && !(fRTCPgs == fRTPgs && dests->rtcpPort.num() == dests->rtpPort.num())) {      fRTCPgs->addDestination(dests->addr, dests->rtcpPort, clientSessionId);    }    if (fRTCPInstance != NULL) {      fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort,          rtcpRRHandler, rtcpRRHandlerClientData);    }  }  if (fRTCPInstance != NULL) {    // Hack: Send an initial RTCP "SR" packet, before the initial RTP packet, so that receivers will (likely) be able to    // get RTCP-synchronized presentation times immediately:    fRTCPInstance->sendReport();  }  if (!fAreCurrentlyPlaying && fMediaSource != NULL) {    if (fRTPSink != NULL) {      fRTPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);      fAreCurrentlyPlaying = True;    } else if (fUDPSink != NULL) {      fUDPSink->startPlaying(*fMediaSource, afterPlayingStreamState, this);      fAreCurrentlyPlaying = True;    }  }}

在这个函数中，首先在 RTCPInstance 还没有创建时去创建它：

RTCPInstance* OnDemandServerMediaSubsession::createRTCP(Groupsock* RTCPgs, unsigned totSessionBW, /* in kbps */    unsigned char const* cname, RTPSink* sink) {  // Default implementation; may be redefined by subclasses:  return RTCPInstance::createNew(envir(), RTCPgs, totSessionBW, cname, sink, NULL/*we're a server*/);}

忽略 RTP/RTCP 包走 TCP 的情况。随后 StreamState::startPlaying() 对 RTP 和 RTCP 的 groupsock 做一些设置，即为它们添加目标地址，并为 RTCPInstance 做了一些设置：

  } else {    // Tell the RTP and RTCP 'groupsocks' about this destination    // (in case they don't already have it):    if (fRTPgs != NULL) fRTPgs->addDestination(dests->addr, dests->rtpPort, clientSessionId);    if (fRTCPgs != NULL && !(fRTCPgs == fRTPgs && dests->rtcpPort.num() == dests->rtpPort.num())) {      fRTCPgs->addDestination(dests->addr, dests->rtcpPort, clientSessionId);    }    if (fRTCPInstance != NULL) {      fRTCPInstance->setSpecificRRHandler(dests->addr.s_addr, dests->rtcpPort,          rtcpRRHandler, rtcpRRHandlerClientData);    }  }

之后 StreamState::startPlaying() 发出一个 RTCP 包。

  if (fRTCPInstance != NULL) {    // Hack: Send an initial RTCP "SR" packet, before the initial RTP packet, so that receivers will (likely) be able to    // get RTCP-synchronized presentation times immediately:    fRTCPInstance->sendReport();  }

fUDPSink 用于流模式为 RAW UDP 的情况，忽略这种流模式的情况。最后执行 MediaSink::startPlaying()，并设置标记 fAreCurrentlyPlaying，表示流播放已经启动。

RTP 包的发送

下面具体来看 RTP 包是怎么被发送出去的。MediaSink::startPlaying() 函数的定义如下：

Boolean MediaSink::startPlaying(MediaSource& source,    afterPlayingFunc* afterFunc,    void* afterClientData) {  // Make sure we're not already being played:  if (fSource != NULL) {    envir().setResultMsg("This sink is already being played");    return False;  }  // Make sure our source is compatible:  if (!sourceIsCompatibleWithUs(source)) {    envir().setResultMsg("MediaSink::startPlaying(): source is not compatible!");    return False;  }  fSource = (FramedSource*)&source;  fAfterFunc = afterFunc;  fAfterClientData = afterClientData;  return continuePlaying();}

在这个函数中，保存了传入的回调及回调的参数，然后执行 continuePlaying()，continuePlaying() 是一个纯虚函数，其实现由 MediaSink 的子类 H264or5VideoRTPSink 实现：

Boolean H264or5VideoRTPSink::continuePlaying() {  // First, check whether we have a 'fragmenter' class set up yet.  // If not, create it now:  if (fOurFragmenter == NULL) {    fOurFragmenter = new H264or5Fragmenter(fHNumber, envir(), fSource, OutPacketBuffer::maxSize,        ourMaxPacketSize() - 12/*RTP hdr size*/);  } else {    fOurFragmenter->reassignInputSource(fSource);  }  fSource = fOurFragmenter;  // Then call the parent class's implementation:  return MultiFramedRTPSink::continuePlaying();}

在这个类中，主要是为 H264or5Fragmenter 设置了流媒体数据源，并将 fSource 设置为 H264or5Fragmenter。在这里，MultiFramedRTPSink 持有的流媒体数据源 FramedSource 由最初在 H264VideoFileServerMediaSubsession 中创建的 H264VideoStreamFramer 变为了 H264or5Fragmenter，而 H264or5Fragmenter 则封装了 H264VideoStreamFramer。

随后 H264or5VideoRTPSink::continuePlaying() 执行 MultiFramedRTPSink::continuePlaying() 做进一步的处理。

Boolean MultiFramedRTPSink::continuePlaying() {  // Send the first packet.  // (This will also schedule any future sends.)  buildAndSendPacket(True);  return True;}. . . . . .void MultiFramedRTPSink::buildAndSendPacket(Boolean isFirstPacket) {  nextTask() = NULL;  fIsFirstPacket = isFirstPacket;  // Set up the RTP header:  unsigned rtpHdr = 0x80000000; // RTP version 2; marker ('M') bit not set (by default; it can be set later)  rtpHdr |= (fRTPPayloadType<<16);  rtpHdr |= fSeqNo; // sequence number  fOutBuf->enqueueWord(rtpHdr);  // Note where the RTP timestamp will go.  // (We can't fill this in until we start packing payload frames.)  fTimestampPosition = fOutBuf->curPacketSize();  fOutBuf->skipBytes(4); // leave a hole for the timestamp  fOutBuf->enqueueWord(SSRC());  // Allow for a special, payload-format-specific header following the  // RTP header:  fSpecialHeaderPosition = fOutBuf->curPacketSize();  fSpecialHeaderSize = specialHeaderSize();  fOutBuf->skipBytes(fSpecialHeaderSize);  // Begin packing as many (complete) frames into the packet as we can:  fTotalFrameSpecificHeaderSizes = 0;  fNoFramesLeft = False;  fNumFramesUsedSoFar = 0;  packFrame();}

MultiFramedRTPSink::continuePlaying() 执行 MultiFramedRTPSink::buildAndSendPacket()。而 MultiFramedRTPSink::buildAndSendPacket() 则是在输出缓冲区构造了 RTP 头部，对于其中暂时无法准确获得的头部字段，还预留了空间。随后调用了 MultiFramedRTPSink::packFrame()。

void MultiFramedRTPSink::packFrame() {  // Get the next frame.  // First, skip over the space we'll use for any frame-specific header:  fCurFrameSpecificHeaderPosition = fOutBuf->curPacketSize();  fCurFrameSpecificHeaderSize = frameSpecificHeaderSize();  fOutBuf->skipBytes(fCurFrameSpecificHeaderSize);  fTotalFrameSpecificHeaderSizes += fCurFrameSpecificHeaderSize;  // See if we have an overflow frame that was too big for the last pkt  if (fOutBuf->haveOverflowData()) {    // Use this frame before reading a new one from the source    unsigned frameSize = fOutBuf->overflowDataSize();    struct timeval presentationTime = fOutBuf->overflowPresentationTime();    unsigned durationInMicroseconds = fOutBuf->overflowDurationInMicroseconds();    fOutBuf->useOverflowData();    afterGettingFrame1(frameSize, 0, presentationTime, durationInMicroseconds);  } else {    // Normal case: we need to read a new frame from the source    if (fSource == NULL) return;    fSource->getNextFrame(fOutBuf->curPtr(), fOutBuf->totalBytesAvailable(),        afterGettingFrame, this, ourHandleClosure, this);  }}

MultiFramedRTPSink::packFrame() 由 FramedSource 的 getNextFrame() 获得帧数据，并在获得帧数据之后得到通知。

void FramedSource::getNextFrame(unsigned char* to, unsigned maxSize,    afterGettingFunc* afterGettingFunc,    void* afterGettingClientData,    onCloseFunc* onCloseFunc,    void* onCloseClientData) {  // Make sure we're not already being read:  if (fIsCurrentlyAwaitingData) {    envir() << "FramedSource[" << this << "]::getNextFrame(): attempting to read more than once at the same time!\n";    envir().internalError();  }  fTo = to;  fMaxSize = maxSize;  fNumTruncatedBytes = 0; // by default; could be changed by doGetNextFrame()  fDurationInMicroseconds = 0; // by default; could be changed by doGetNextFrame()  fAfterGettingFunc = afterGettingFunc;  fAfterGettingClientData = afterGettingClientData;  fOnCloseFunc = onCloseFunc;  fOnCloseClientData = onCloseClientData;  fIsCurrentlyAwaitingData = True;  doGetNextFrame();}

这个函数主要用于为 FramedSource 设置媒体流数据要读到哪里，可以读多少自己，以及回调函数的地址。并最终执行 doGetNextFrame() 读取数据。

最终数据将由 ByteStreamFileSource 的 doGetNextFrame() 执行读取任务的调度，并从文件中读取。

#0  ByteStreamFileSource::doGetNextFrame (this=0x6d8f10) at ByteStreamFileSource.cpp:96#1  0x000000000043004c in FramedSource::getNextFrame (this=0x6d8f10, to=0x6da9c0 "(\243\203\367\377\177", maxSize=150000,     afterGettingFunc=0x46f6c8 <StreamParser::afterGettingBytes(void*, unsigned int, unsigned int, timeval, unsigned int)>,     afterGettingClientData=0x6d91b0, onCloseFunc=0x46f852 <StreamParser::onInputClosure(void*)>, onCloseClientData=0x6d91b0) at FramedSource.cpp:78-------------------------------------------------------------------------------------------------------------------------------------#2  0x000000000046f69c in StreamParser::ensureValidBytes1 (this=0x6d91b0, numBytesNeeded=4) at StreamParser.cpp:159#3  0x00000000004343e5 in StreamParser::ensureValidBytes (this=0x6d91b0, numBytesNeeded=4) at StreamParser.hh:118#4  0x0000000000434179 in StreamParser::test4Bytes (this=0x6d91b0) at StreamParser.hh:54#5  0x0000000000471b85 in H264or5VideoStreamParser::parse (this=0x6d91b0) at H264or5VideoStreamFramer.cpp:951#6  0x000000000043510f in MPEGVideoStreamFramer::continueReadProcessing (this=0x6d9000) at MPEGVideoStreamFramer.cpp:159#7  0x0000000000435077 in MPEGVideoStreamFramer::doGetNextFrame (this=0x6d9000) at MPEGVideoStreamFramer.cpp:142#8  0x000000000043004c in FramedSource::getNextFrame (this=0x6d9000, to=0x748d61 "", maxSize=100000,     afterGettingFunc=0x474cd2 <H264or5Fragmenter::afterGettingFrame(void*, unsigned int, unsigned int, timeval, unsigned int)>,     afterGettingClientData=0x700300, onCloseFunc=0x4300c6 <FramedSource::handleClosure(void*)>, onCloseClientData=0x700300) at FramedSource.cpp:78-------------------------------------------------------------------------------------------------------------------------------------#9  0x000000000047480a in H264or5Fragmenter::doGetNextFrame (this=0x700300) at H264or5VideoRTPSink.cpp:181#10 0x000000000043004c in FramedSource::getNextFrame (this=0x700300, to=0x7304ec "", maxSize=100452,     afterGettingFunc=0x45af82 <MultiFramedRTPSink::afterGettingFrame(void*, unsigned int, unsigned int, timeval, unsigned int)>,     afterGettingClientData=0x6d92e0, onCloseFunc=0x45b96c <MultiFramedRTPSink::ourHandleClosure(void*)>, onCloseClientData=0x6d92e0) at FramedSource.cpp:78-------------------------------------------------------------------------------------------------------------------------------------#11 0x000000000045af61 in MultiFramedRTPSink::packFrame (this=0x6d92e0) at MultiFramedRTPSink.cpp:224#12 0x000000000045adae in MultiFramedRTPSink::buildAndSendPacket (this=0x6d92e0, isFirstPacket=1 '\001') at MultiFramedRTPSink.cpp:199#13 0x000000000045abed in MultiFramedRTPSink::continuePlaying (this=0x6d92e0) at MultiFramedRTPSink.cpp:159-------------------------------------------------------------------------------------------------------------------------------------#14 0x000000000047452a in H264or5VideoRTPSink::continuePlaying (this=0x6d92e0) at H264or5VideoRTPSink.cpp:127#15 0x0000000000405d2a in MediaSink::startPlaying (this=0x6d92e0, source=..., afterFunc=0x4621f4 <afterPlayingStreamState(void*)>,     afterClientData=0x6d95b0) at MediaSink.cpp:78#16 0x00000000004626ea in StreamState::startPlaying (this=0x6d95b0, dests=0x6d9620, clientSessionId=1584618840,     rtcpRRHandler=0x407280 <GenericMediaServer::ClientSession::noteClientLiveness(GenericMediaServer::ClientSession*)>, rtcpRRHandlerClientData=0x70ba40,     serverRequestAlternativeByteHandler=0x4093a6 <RTSPServer::RTSPClientConnection::handleAlternativeRequestByte(void*, unsigned char)>,     serverRequestAlternativeByteHandlerClientData=0x6ce910) at OnDemandServerMediaSubsession.cpp:576#17 0x000000000046138d in OnDemandServerMediaSubsession::startStream (this=0x6d8710, clientSessionId=1584618840, streamToken=0x6d95b0,     rtcpRRHandler=0x407280 <GenericMediaServer::ClientSession::noteClientLiveness(GenericMediaServer::ClientSession*)>, rtcpRRHandlerClientData=0x70ba40,     rtpSeqNum=@0x7fffffffcd76: 0, rtpTimestamp=@0x7fffffffcdc0: 0,     serverRequestAlternativeByteHandler=0x4093a6 <RTSPServer::RTSPClientConnection::handleAlternativeRequestByte(void*, unsigned char)>,     serverRequestAlternativeByteHandlerClientData=0x6ce910) at OnDemandServerMediaSubsession.cpp:223

这个调用栈比较深。看起来可能会让人感觉比较费解。实际上 live555 中采用装饰器模式来设计 FramedSource，一个 FramedSource 可以包装另一个 FramedSource，并额外提供一些功能，或为了性能优化，或为了数据解析等。

live555 中众多的 FramedSource 类之间的关系大概如下图所示：

上面的调用栈，也主要根据 FramedSource 的包装关系，由虚线分割为几个不同的阶段。

在 ByteStreamFileSource 的 doGetNextFrame() 中，调度读取任务：

void ByteStreamFileSource::doGetNextFrame() {  if (feof(fFid) || ferror(fFid) || (fLimitNumBytesToStream && fNumBytesToStream == 0)) {    handleClosure();    return;  }#ifdef READ_FROM_FILES_SYNCHRONOUSLY  doReadFromFile();#else  if (!fHaveStartedReading) {    // Await readable data from the file:    envir().taskScheduler().turnOnBackgroundReadHandling(fileno(fFid),           (TaskScheduler::BackgroundHandlerProc*)&fileReadableHandler, this);    fHaveStartedReading = True;  }#endif}

ByteStreamFileSource::fileReadableHandler() 读取流媒体内容，并通知调用者：

void FramedSource::afterGetting(FramedSource* source) {  source->nextTask() = NULL;  source->fIsCurrentlyAwaitingData = False;  // indicates that we can be read again  // Note that this needs to be done here, in case the "fAfterFunc"  // called below tries to read another frame (which it usually will)  if (source->fAfterGettingFunc != NULL) {    (*(source->fAfterGettingFunc))(source->fAfterGettingClientData,        source->fFrameSize, source->fNumTruncatedBytes,        source->fPresentationTime,        source->fDurationInMicroseconds);  }}. . . . . .void ByteStreamFileSource::fileReadableHandler(ByteStreamFileSource* source, int /*mask*/) {  if (!source->isCurrentlyAwaitingData()) {    source->doStopGettingFrames(); // we're not ready for the data yet    return;  }  source->doReadFromFile();}void ByteStreamFileSource::doReadFromFile() {  // Try to read as many bytes as will fit in the buffer provided (or "fPreferredFrameSize" if less)  if (fLimitNumBytesToStream && fNumBytesToStream < (u_int64_t)fMaxSize) {    fMaxSize = (unsigned)fNumBytesToStream;  }  if (fPreferredFrameSize > 0 && fPreferredFrameSize < fMaxSize) {    fMaxSize = fPreferredFrameSize;  }#ifdef READ_FROM_FILES_SYNCHRONOUSLY  fFrameSize = fread(fTo, 1, fMaxSize, fFid);#else  if (fFidIsSeekable) {    fFrameSize = fread(fTo, 1, fMaxSize, fFid);  } else {    // For non-seekable files (e.g., pipes), call "read()" rather than "fread()", to ensure that the read doesn't block:    fFrameSize = read(fileno(fFid), fTo, fMaxSize);  }#endif  if (fFrameSize == 0) {    handleClosure();    return;  }  fNumBytesToStream -= fFrameSize;  // Set the 'presentation time':  if (fPlayTimePerFrame > 0 && fPreferredFrameSize > 0) {    if (fPresentationTime.tv_sec == 0 && fPresentationTime.tv_usec == 0) {      // This is the first frame, so use the current time:      gettimeofday(&fPresentationTime, NULL);    } else {      // Increment by the play time of the previous data:      unsigned uSeconds = fPresentationTime.tv_usec + fLastPlayTime;      fPresentationTime.tv_sec += uSeconds/1000000;      fPresentationTime.tv_usec = uSeconds%1000000;    }    // Remember the play time of this data:    fLastPlayTime = (fPlayTimePerFrame*fFrameSize)/fPreferredFrameSize;    fDurationInMicroseconds = fLastPlayTime;  } else {    // We don't know a specific play time duration for this data,    // so just record the current time as being the 'presentation time':    gettimeofday(&fPresentationTime, NULL);  }  // Inform the reader that he has data:#ifdef READ_FROM_FILES_SYNCHRONOUSLY  // To avoid possible infinite recursion, we need to return to the event loop to do this:  nextTask() = envir().taskScheduler().scheduleDelayedTask(0,                (TaskFunc*)FramedSource::afterGetting, this);#else  // Because the file read was done from the event loop, we can call the  // 'after getting' function directly, without risk of infinite recursion:  FramedSource::afterGetting(this);#endif}

数据读取完成之后，MultiFramedRTPSink 将得到通知：

#0  MultiFramedRTPSink::afterGettingFrame (clientData=0x6d92e0, numBytesRead=18, numTruncatedBytes=0, presentationTime=...,     durationInMicroseconds=0) at MultiFramedRTPSink.cpp:233---------------------------------------------------------------------------------------------------------------------------#1  0x00000000004300c2 in FramedSource::afterGetting (source=0x7002c0) at FramedSource.cpp:92#2  0x0000000000474ca6 in H264or5Fragmenter::doGetNextFrame (this=0x7002c0) at H264or5VideoRTPSink.cpp:263#3  0x0000000000474dac in H264or5Fragmenter::afterGettingFrame1 (this=0x7002c0, frameSize=18, numTruncatedBytes=0, presentationTime=...,     durationInMicroseconds=0) at H264or5VideoRTPSink.cpp:292#4  0x0000000000474d25 in H264or5Fragmenter::afterGettingFrame (clientData=0x7002c0, frameSize=18, numTruncatedBytes=0, presentationTime=...,     durationInMicroseconds=0) at H264or5VideoRTPSink.cpp:279---------------------------------------------------------------------------------------------------------------------------#5  0x00000000004300c2 in FramedSource::afterGetting (source=0x6d9000) at FramedSource.cpp:92#6  0x00000000004351ea in MPEGVideoStreamFramer::continueReadProcessing (this=0x6d9000) at MPEGVideoStreamFramer.cpp:179#7  0x00000000004350da in MPEGVideoStreamFramer::continueReadProcessing (clientData=0x6d9000) at MPEGVideoStreamFramer.cpp:155#8  0x000000000046f84f in StreamParser::afterGettingBytes1 (this=0x6d91b0, numBytesRead=150000, presentationTime=...) at StreamParser.cpp:191#9  0x000000000046f718 in StreamParser::afterGettingBytes (clientData=0x6d91b0, numBytesRead=150000, presentationTime=...)    at StreamParser.cpp:170---------------------------------------------------------------------------------------------------------------------------#10 0x00000000004300c2 in FramedSource::afterGetting (source=0x6d8f10) at FramedSource.cpp:92#11 0x0000000000430c2c in ByteStreamFileSource::doReadFromFile (this=0x6d8f10) at ByteStreamFileSource.cpp:182#12 0x00000000004309cb in ByteStreamFileSource::fileReadableHandler (source=0x6d8f10) at ByteStreamFileSource.cpp:126

我们同样将回调的调用栈，根据 FramedSource 的包装关系，分为几个阶段，不同阶段以虚线分割。

MultiFramedRTPSink::afterGettingFrame() 函数定义如下：

void MultiFramedRTPSink::afterGettingFrame(void* clientData, unsigned numBytesRead,            unsigned numTruncatedBytes,            struct timeval presentationTime,            unsigned durationInMicroseconds) {  MultiFramedRTPSink* sink = (MultiFramedRTPSink*)clientData;  sink->afterGettingFrame1(numBytesRead, numTruncatedBytes,               presentationTime, durationInMicroseconds);}

在这个函数中调用 afterGettingFrame1()， afterGettingFrame1() 则会根据需要调用 sendPacketIfNecessary()。MultiFramedRTPSink::sendPacketIfNecessary() 定义如下：

void MultiFramedRTPSink::sendPacketIfNecessary() {  if (fNumFramesUsedSoFar > 0) {    // Send the packet:#ifdef TEST_LOSS    if ((our_random()%10) != 0) // simulate 10% packet loss ######endif    if (!fRTPInterface.sendPacket(fOutBuf->packet(), fOutBuf->curPacketSize())) {      // if failure handler has been specified, call it      if (fOnSendErrorFunc != NULL) (*fOnSendErrorFunc)(fOnSendErrorData);    }    ++fPacketCount;    fTotalOctetCount += fOutBuf->curPacketSize();    fOctetCount += fOutBuf->curPacketSize()      - rtpHeaderSize - fSpecialHeaderSize - fTotalFrameSpecificHeaderSizes;    ++fSeqNo; // for next time  }  if (fOutBuf->haveOverflowData()      && fOutBuf->totalBytesAvailable() > fOutBuf->totalBufferSize()/2) {    // Efficiency hack: Reset the packet start pointer to just in front of    // the overflow data (allowing for the RTP header and special headers),    // so that we probably don't have to "memmove()" the overflow data    // into place when building the next packet:    unsigned newPacketStart = fOutBuf->curPacketSize()      - (rtpHeaderSize + fSpecialHeaderSize + frameSpecificHeaderSize());    fOutBuf->adjustPacketStart(newPacketStart);  } else {    // Normal case: Reset the packet start pointer back to the start:    fOutBuf->resetPacketStart();  }  fOutBuf->resetOffset();  fNumFramesUsedSoFar = 0;  if (fNoFramesLeft) {    // We're done:    onSourceClosure();  } else {    // We have more frames left to send.  Figure out when the next frame    // is due to start playing, then make sure that we wait this long before    // sending the next packet.    struct timeval timeNow;    gettimeofday(&timeNow, NULL);    int secsDiff = fNextSendTime.tv_sec - timeNow.tv_sec;    int64_t uSecondsToGo = secsDiff*1000000 + (fNextSendTime.tv_usec - timeNow.tv_usec);    if (uSecondsToGo < 0 || secsDiff < 0) { // sanity check: Make sure that the time-to-delay is non-negative:      uSecondsToGo = 0;    }    // Delay this amount of time:    nextTask() = envir().taskScheduler().scheduleDelayedTask(uSecondsToGo, (TaskFunc*)sendNext, this);  }}

在 MultiFramedRTPSink::sendPacketIfNecessary() 中，会发送帧数据。且如果流媒体数据发送没有结束的话，在一帧数据发送完成之后，会调度一个定时器任务 MultiFramedRTPSink::sendNext() 再次发送帧数据。

MultiFramedRTPSink::sendNext() 执行与 MultiFramedRTPSink::continuePlaying() 类似的流程，获取下一帧数据并发送。

void MultiFramedRTPSink::sendNext(void* firstArg) {  MultiFramedRTPSink* sink = (MultiFramedRTPSink*)firstArg;  sink->buildAndSendPacket(False);}

当然也并不是每一次发送帧数据的时候，都需要直接从流媒体源中去获得数据。在 StreamParser 中会做判断，当需要帧数据的时候，它会发起对流媒体文件的读取。若无需从文件中读取流媒体数据，则会直接回调：

#0  MultiFramedRTPSink::sendPacketIfNecessary (this=0x702140) at MultiFramedRTPSink.cpp:365#1  0x000000000045b5a4 in MultiFramedRTPSink::afterGettingFrame1 (this=0x702140, frameSize=1444, numTruncatedBytes=0, presentationTime=...,     durationInMicroseconds=40000) at MultiFramedRTPSink.cpp:347#2  0x000000000045afd5 in MultiFramedRTPSink::afterGettingFrame (clientData=0x702140, numBytesRead=1444, numTruncatedBytes=0,     presentationTime=..., durationInMicroseconds=40000) at MultiFramedRTPSink.cpp:235#3  0x00000000004300c2 in FramedSource::afterGetting (source=0x7036d0) at FramedSource.cpp:92------------------------------------------------------------------------------------------------------------------------------------#4  0x0000000000474ca6 in H264or5Fragmenter::doGetNextFrame (this=0x7036d0) at H264or5VideoRTPSink.cpp:263#5  0x0000000000474dac in H264or5Fragmenter::afterGettingFrame1 (this=0x7036d0, frameSize=53527, numTruncatedBytes=0, presentationTime=...,     durationInMicroseconds=40000) at H264or5VideoRTPSink.cpp:292#6  0x0000000000474d25 in H264or5Fragmenter::afterGettingFrame (clientData=0x7036d0, frameSize=53527, numTruncatedBytes=0,     presentationTime=..., durationInMicroseconds=40000) at H264or5VideoRTPSink.cpp:279#7  0x00000000004300c2 in FramedSource::afterGetting (source=0x701e20) at FramedSource.cpp:92------------------------------------------------------------------------------------------------------------------------------------#8  0x00000000004351ea in MPEGVideoStreamFramer::continueReadProcessing (this=0x701e20) at MPEGVideoStreamFramer.cpp:179#9  0x0000000000435077 in MPEGVideoStreamFramer::doGetNextFrame (this=0x701e20) at MPEGVideoStreamFramer.cpp:142------------------------------------------------------------------------------------------------------------------------------------#10 0x000000000043004c in FramedSource::getNextFrame (this=0x701e20, to=0x7c3091 "\205\270@\367\017\204?\017", <incomplete sequence \340>,     maxSize=100000,     afterGettingFunc=0x474cd2 <H264or5Fragmenter::afterGettingFrame(void*, unsigned int, unsigned int, timeval, unsigned int)>,     afterGettingClientData=0x7036d0, onCloseFunc=0x4300c6 <FramedSource::handleClosure(void*)>, onCloseClientData=0x7036d0)    at FramedSource.cpp:78#11 0x000000000047480a in H264or5Fragmenter::doGetNextFrame (this=0x7036d0) at H264or5VideoRTPSink.cpp:181------------------------------------------------------------------------------------------------------------------------------------#12 0x000000000043004c in FramedSource::getNextFrame (this=0x7036d0, to=0x7aa81c "|\205\270@\367\017\204?\017", <incomplete sequence \340>,     maxSize=100452,     afterGettingFunc=0x45af82 <MultiFramedRTPSink::afterGettingFrame(void*, unsigned int, unsigned int, timeval, unsigned int)>,     afterGettingClientData=0x702140, onCloseFunc=0x45b96c <MultiFramedRTPSink::ourHandleClosure(void*)>, onCloseClientData=0x702140)    at FramedSource.cpp:78#13 0x000000000045af61 in MultiFramedRTPSink::packFrame (this=0x702140) at MultiFramedRTPSink.cpp:224#14 0x000000000045adae in MultiFramedRTPSink::buildAndSendPacket (this=0x702140, isFirstPacket=0 '\000') at MultiFramedRTPSink.cpp:199#15 0x000000000045b969 in MultiFramedRTPSink::sendNext (firstArg=0x702140) at MultiFramedRTPSink.cpp:422#16 0x000000000047f165 in AlarmHandler::handleTimeout (this=0x7038a0) at BasicTaskScheduler0.cpp:34#17 0x000000000047d268 in DelayQueue::handleAlarm (this=0x6cdc28) at DelayQueue.cpp:187#18 0x000000000047c196 in BasicTaskScheduler::SingleStep (this=0x6cdc20, maxDelayTime=0) at BasicTaskScheduler.cpp:212

总结一下 RTP 数据包的发送过程：

1. OnDemandServerMediaSubsession 中执行 startStream() 时，将发起一个对流媒体文件进行读取的任务，读取文件的工作由 ByteStreamFileSource 的 doReadFromFile() 执行。
2. 在文件读取了一些数据之后，MultiFramedRTPSink 得到回调 afterGetting()，在这个回调中，发送帧数据。
3. MultiFramedRTPSink 的回调中，如果流媒体数据还没有读完的话，则调度一个定时器任务，一段时间之后再次发起获取帧数据的动作。
4. 重复 2 和 3 两步，直到所有的数据都发送完。

RTCP 包的接收

StreamState::startPlaying() 通过 OnDemandServerMediaSubsession::createRTCP() 创建 RTCPInstance：

RTCPInstance* OnDemandServerMediaSubsession::createRTCP(Groupsock* RTCPgs, unsigned totSessionBW, /* in kbps */    unsigned char const* cname, RTPSink* sink) {  fprintf(stderr, "OnDemandServerMediaSubsession::createRTCP().\n");  // Default implementation; may be redefined by subclasses:  return RTCPInstance::createNew(envir(), RTCPgs, totSessionBW, cname, sink, NULL/*we're a server*/);}

OnDemandServerMediaSubsession::createRTCP() 则通过 RTCPInstance::createNew() 创建：

RTCPInstance::RTCPInstance(UsageEnvironment& env, Groupsock* RTCPgs,               unsigned totSessionBW,               unsigned char const* cname,               RTPSink* sink, RTPSource* source,               Boolean isSSMSource)  : Medium(env), fRTCPInterface(this, RTCPgs), fTotSessionBW(totSessionBW),    fSink(sink), fSource(source), fIsSSMSource(isSSMSource),    fCNAME(RTCP_SDES_CNAME, cname), fOutgoingReportCount(1),    fAveRTCPSize(0), fIsInitial(1), fPrevNumMembers(0),    fLastSentSize(0), fLastReceivedSize(0), fLastReceivedSSRC(0),    fTypeOfEvent(EVENT_UNKNOWN), fTypeOfPacket(PACKET_UNKNOWN_TYPE),    fHaveJustSentPacket(False), fLastPacketSentSize(0),    fByeHandlerTask(NULL), fByeHandlerClientData(NULL),    fSRHandlerTask(NULL), fSRHandlerClientData(NULL),    fRRHandlerTask(NULL), fRRHandlerClientData(NULL),    fSpecificRRHandlerTable(NULL),    fAppHandlerTask(NULL), fAppHandlerClientData(NULL) {#ifdef DEBUG  fprintf(stderr, "RTCPInstance[%p]::RTCPInstance()\n", this);#endif  if (fTotSessionBW == 0) { // not allowed!    env << "RTCPInstance::RTCPInstance error: totSessionBW parameter should not be zero!\n";    fTotSessionBW = 1;  }  if (isSSMSource) RTCPgs->multicastSendOnly(); // don't receive multicast  double timeNow = dTimeNow();  fPrevReportTime = fNextReportTime = timeNow;  fKnownMembers = new RTCPMemberDatabase(*this);  fInBuf = new unsigned char[maxRTCPPacketSize];  if (fKnownMembers == NULL || fInBuf == NULL) return;  fNumBytesAlreadyRead = 0;  fOutBuf = new OutPacketBuffer(preferredRTCPPacketSize, maxRTCPPacketSize, maxRTCPPacketSize);  if (fOutBuf == NULL) return;  if (fSource != NULL && fSource->RTPgs() == RTCPgs) {    // We're receiving RTCP reports that are multiplexed with RTP, so ask the RTP source    // to give them to us:    fSource->registerForMultiplexedRTCPPackets(this);  } else {    // Arrange to handle incoming reports from the network:    TaskScheduler::BackgroundHandlerProc* handler      = (TaskScheduler::BackgroundHandlerProc*)&incomingReportHandler;    fRTCPInterface.startNetworkReading(handler);  }  // Send our first report.  fTypeOfEvent = EVENT_REPORT;  onExpire(this);}. . . . . .RTCPInstance* RTCPInstance::createNew(UsageEnvironment& env, Groupsock* RTCPgs,                      unsigned totSessionBW,                      unsigned char const* cname,                      RTPSink* sink, RTPSource* source,                      Boolean isSSMSource) {  return new RTCPInstance(env, RTCPgs, totSessionBW, cname, sink, source,              isSSMSource);}

可以看到，在 RTCPInstance 的构造函数中，调用 RTPInterface::startNetworkReading() 注册了一个回调：

void RTPInterface::startNetworkReading(TaskScheduler::BackgroundHandlerProc* handlerProc) {  // Normal case: Arrange to read UDP packets:  envir().taskScheduler().    turnOnBackgroundReadHandling(fGS->socketNum(), handlerProc, fOwner);  // Also, receive RTP over TCP, on each of our TCP connections:  fReadHandlerProc = handlerProc;  for (tcpStreamRecord* streams = fTCPStreams; streams != NULL;       streams = streams->fNext) {    // Get a socket descriptor for "streams->fStreamSocketNum":    SocketDescriptor* socketDescriptor = lookupSocketDescriptor(envir(), streams->fStreamSocketNum);    // Tell it about our subChannel:    socketDescriptor->registerRTPInterface(streams->fStreamChannelId, this);  }}

在 RTPInterface::startNetworkReading() 中则会向 TaskScheduler 注册 RTCP 的 socket 及该 socket 上的事件的处理程序。live555 中正是通过这种方式，在有 RTCP 包到来时得到通知，并通过 RTCPInstance::incomingReportHandler() 来处理 RTCP 包的。

RTCP 包的发送

RTCP 包根据需要，由 RTCPInstance::sendReport() 等函数发送：

void RTCPInstance::sendReport() {#ifdef DEBUG  fprintf(stderr, "sending REPORT\n");#endif  // Begin by including a SR and/or RR report:  if (!addReport()) return;  // Then, include a SDES:  addSDES();  // Send the report:  sendBuiltPacket();  // Periodically clean out old members from our SSRC membership database:  const unsigned membershipReapPeriod = 5;  if ((++fOutgoingReportCount) % membershipReapPeriod == 0) {    unsigned threshold = fOutgoingReportCount - membershipReapPeriod;    fKnownMembers->reapOldMembers(threshold);  }}void RTCPInstance::sendBYE() {#ifdef DEBUG  fprintf(stderr, "sending BYE\n");#endif  // The packet must begin with a SR and/or RR report:  (void)addReport(True);  addBYE();  sendBuiltPacket();}void RTCPInstance::sendBuiltPacket() {#ifdef DEBUG  fprintf(stderr, "sending RTCP packet\n");  unsigned char* p = fOutBuf->packet();  for (unsigned i = 0; i < fOutBuf->curPacketSize(); ++i) {    if (i%4 == 0) fprintf(stderr," ");    fprintf(stderr, "%02x", p[i]);  }  fprintf(stderr, "\n");#endif  unsigned reportSize = fOutBuf->curPacketSize();  fRTCPInterface.sendPacket(fOutBuf->packet(), reportSize);  fOutBuf->resetOffset();  fLastSentSize = IP_UDP_HDR_SIZE + reportSize;  fHaveJustSentPacket = True;  fLastPacketSentSize = reportSize;}

就像在 StreamState::startPlaying() 中看到的那样。

Done.

live555 源码分析系列文章

live555 源码分析：简介
live555 源码分析：基础设施
live555 源码分析：MediaSever
Wireshark 抓包分析 RTSP/RTP/RTCP 基本工作过程
live555 源码分析：RTSPServer
live555 源码分析：DESCRIBE 的处理
live555 源码分析：SETUP 的处理
live555 源码分析：PLAY 的处理
live555 源码分析：RTSPServer 组件结构
live555 源码分析：ServerMediaSession
live555 源码分析：子会话 SDP 行生成
live555 源码分析：子会话 SETUP
live555 源码分析：播放启动