openRtsp分析4

在上篇中分析了setup建立的过程，现在再来看看sink建立的过程。continueAfterSETUP之后直接就进入到了setupStreams。这里根据subsession->codecName()建立对应的fileSink。

 fileSink = H264VideoFileSink::createNew(*env, outFileName,
						  subsession->fmtp_spropparametersets(),
						  fileSinkBufferSize, oneFilePerFrame);

这里以h264为例。fileSink创建的过程就一一分析，接着往下看。

subsession->sink->startPlaying(*(subsession->readSource()),
					 subsessionAfterPlaying,
					 subsession);

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();
}

这里的fSource就是fReadSource，也就是在上一篇中建立的H264VideoRTPSource。这里还做了其他一些赋值，接着就是执行continuePlaying在MediaSink中为虚函数，但是在FileSink中定义。

Boolean FileSink::continuePlaying() {
  if (fSource == NULL) return False;

  fSource->getNextFrame(fBuffer, fBufferSize,
			afterGettingFrame, this,
			onSourceClosure, this);

  return True;
}

getNextFrame在FramedSource中定义，主要是把fBuffer赋值给fTo，把afterGettingFrame赋给fAfterGettingFrame。之后就是执行doGetNextFrame，这个在H264VideoRTPSource的父类MultiFrameRTPSource中定义。

void MultiFramedRTPSource::doGetNextFrame() {
  if (!fAreDoingNetworkReads) {
    // Turn on background read handling of incoming packets:
    fAreDoingNetworkReads = True;
    TaskScheduler::BackgroundHandlerProc* handler
      = (TaskScheduler::BackgroundHandlerProc*)&networkReadHandler;
    fRTPInterface.startNetworkReading(handler);
  }

  fSavedTo = fTo;
  fSavedMaxSize = fMaxSize;
  fFrameSize = 0; // for now
  fNeedDelivery = True;
  doGetNextFrame1();
}

这里的fRTPInterface是在建立H264VideoRTPSource在初始化父类RTPSource时初始化的，

RTPSource::RTPSource(UsageEnvironment& env, Groupsock* RTPgs,
		     unsigned char rtpPayloadFormat,
		     u_int32_t rtpTimestampFrequency)
  : FramedSource(env),
    fRTPInterface(this, RTPgs),
    fCurPacketHasBeenSynchronizedUsingRTCP(False),
    fLastReceivedSSRC(0),
    fRTPPayloadFormat(rtpPayloadFormat),
    fTimestampFrequency(rtpTimestampFrequency),
    fSSRC(our_random32()) {
  fReceptionStatsDB = new RTPReceptionStatsDB();
}

这里的参数RTPgs是fRTPSocket，来看看他的startNetWorkReading

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);
  }
}

这里的turnOnBackgroundReadHandling在UsageEnvironment中定义

void turnOnBackgroundReadHandling(int socketNum, BackgroundHandlerProc* handlerProc, void* clientData) {
    setBackgroundHandling(socketNum, SOCKET_READABLE, handlerProc, clientData);
  }

但是setBackgroundHandling在UsageEnvironment中为纯虚函数，在BasicTaskScheduler中有定义在前面的分析见过这个方法，这里就是将SOCKET_READABLE加入到

socket的set集中，然后将handlerProc即networkReadHandler加入到fHandlers中，这样在eventloop中的singleSetup的时候有在socketNum上的read就会监听到。而这里的fGS->socketNum()就是H264VideoRTPSource的时候传入的，即在上一篇inital的时候创建的。

我们再来看看rtp数据接收到后处理的方法。

void MultiFramedRTPSource::networkReadHandler(MultiFramedRTPSource* source, int /*mask*/) {
  source->networkReadHandler1();
}

可以看到networkReadHandler1中主要是对rtp的拆包处理

void MultiFramedRTPSource::networkReadHandler1() {
  BufferedPacket* bPacket = fPacketReadInProgress;
  if (bPacket == NULL) {
    // Normal case: Get a free BufferedPacket descriptor to hold the new network packet:
    bPacket = fReorderingBuffer->getFreePacket(this);
  }

  // Read the network packet, and perform sanity checks on the RTP header:
  Boolean readSuccess = False;
  do {
    Boolean packetReadWasIncomplete = fPacketReadInProgress != NULL;
    if (!bPacket->fillInData(fRTPInterface, packetReadWasIncomplete)) break;
    if (packetReadWasIncomplete) {
      // We need additional read(s) before we can process the incoming packet:
      fPacketReadInProgress = bPacket;
      return;
    } else {
      fPacketReadInProgress = NULL;
    }
#ifdef TEST_LOSS
    setPacketReorderingThresholdTime(0);
       // don't wait for 'lost' packets to arrive out-of-order later
    if ((our_random()%10) == 0) break; // simulate 10% packet loss
#endif

    // Check for the 12-byte RTP header:
    if (bPacket->dataSize() < 12) break;
    unsigned rtpHdr = ntohl(*(u_int32_t*)(bPacket->data())); ADVANCE(4);
    Boolean rtpMarkerBit = (rtpHdr&0x00800000) != 0;
    unsigned rtpTimestamp = ntohl(*(u_int32_t*)(bPacket->data()));ADVANCE(4);
    unsigned rtpSSRC = ntohl(*(u_int32_t*)(bPacket->data())); ADVANCE(4);

    // Check the RTP version number (it should be 2):
    if ((rtpHdr&0xC0000000) != 0x80000000) break;

    // Skip over any CSRC identifiers in the header:
    unsigned cc = (rtpHdr>>24)&0xF;
    if (bPacket->dataSize() < cc) break;
    ADVANCE(cc*4);

    // Check for (& ignore) any RTP header extension
    if (rtpHdr&0x10000000) {
      if (bPacket->dataSize() < 4) break;
      unsigned extHdr = ntohl(*(u_int32_t*)(bPacket->data())); ADVANCE(4);
      unsigned remExtSize = 4*(extHdr&0xFFFF);
      if (bPacket->dataSize() < remExtSize) break;
      ADVANCE(remExtSize);
    }

    // Discard any padding bytes:
    if (rtpHdr&0x20000000) {
      if (bPacket->dataSize() == 0) break;
      unsigned numPaddingBytes
	= (unsigned)(bPacket->data())[bPacket->dataSize()-1];
      if (bPacket->dataSize() < numPaddingBytes) break;
      bPacket->removePadding(numPaddingBytes);
    }
    // Check the Payload Type.
    if ((unsigned char)((rtpHdr&0x007F0000)>>16)
	!= rtpPayloadFormat()) {
      break;
    }

    // The rest of the packet is the usable data.  Record and save it:
    if (rtpSSRC != fLastReceivedSSRC) {
      // The SSRC of incoming packets has changed.  Unfortunately we don't yet handle streams that contain multiple SSRCs,
      // but we can handle a single-SSRC stream where the SSRC changes occasionally:
      fLastReceivedSSRC = rtpSSRC;
      fReorderingBuffer->resetHaveSeenFirstPacket();
    }
    unsigned short rtpSeqNo = (unsigned short)(rtpHdr&0xFFFF);
    Boolean usableInJitterCalculation
      = packetIsUsableInJitterCalculation((bPacket->data()),
						  bPacket->dataSize());
    struct timeval presentationTime; // computed by:
    Boolean hasBeenSyncedUsingRTCP; // computed by:
    receptionStatsDB()
      .noteIncomingPacket(rtpSSRC, rtpSeqNo, rtpTimestamp,
			  timestampFrequency(),
			  usableInJitterCalculation, presentationTime,
			  hasBeenSyncedUsingRTCP, bPacket->dataSize());

    // Fill in the rest of the packet descriptor, and store it:
    struct timeval timeNow;
    gettimeofday(&timeNow, NULL);
    bPacket->assignMiscParams(rtpSeqNo, rtpTimestamp, presentationTime,
			      hasBeenSyncedUsingRTCP, rtpMarkerBit,
			      timeNow);
    if (!fReorderingBuffer->storePacket(bPacket)) break;

    readSuccess = True;
  } while (0);
  if (!readSuccess) fReorderingBuffer->freePacket(bPacket);

  doGetNextFrame1();
  // If we didn't get proper data this time, we'll get another chance
}

分析玩rtp头等工作后就是doGetNextFrame1