live555从RTSP服务器读取数据到使用接收到的数据流程分析

本文在linux环境下编译live555工程，并用cgdb调试工具对live555工程中的testProgs目录下的openRTSP的执行过程进行了跟踪分析，直到将从socket端读取视频数据并保存为对应的视频和音频数据为止。

进入testProgs目录，执行./openRTSP rtsp://xxxx/test.mp4

对于RTSP协议的处理部分，可设置断点在setupStreams函数中，并跟踪即可进行分析。

这里主要分析进入如下的while(1)循环中的代码

void BasicTaskScheduler0::doEventLoop(char* watchVariable) {  // Repeatedly loop, handling readble sockets and timed events:  while (1)   {    if (watchVariable != NULL && *watchVariable != 0) break;    SingleStep();  }}

 

从这里可知，live555在客户端处理数据实际上是单线程的程序，不断执行SingleStep()函数中的代码。通过查看该函数代码里，下面一句代码为重点

 (*handler->handlerProc)(handler->clientData, resultConditionSet);

 

其中该条代码出现了两次，通过调试跟踪它的执行轨迹，第一次出现调用的函数是为了处理和RTSP服务器的通信协议的商定，而第二次出现调用的函数才是处理真正的视频和音频数据。对于RTSP通信协议的分析我们暂且不讨论，而直接进入第二次调用该函数的部分。

在我们的调试过程中在执行到上面的函数时就直接调用到livemedia目录下的如下函数

 

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

//下面这个函数实现的主要功能就是从socket端读取数据并存储数据

 

void MultiFramedRTPSource::networkReadHandler1() {  BufferedPacket* bPacket = fPacketReadInProgress;  if (bPacket == NULL)  {    // Normal case: Get a free BufferedPacket descriptor to hold the new network packet:    //分配一块新的存储空间来存储从socket端读取的数据    bPacket = fReorderingBuffer->getFreePacket(this);  }  // Read the network packet, and perform sanity checks on the RTP header:  Boolean readSuccess = False;  do   {    Boolean packetReadWasIncomplete = fPacketReadInProgress != NULL;    //fillInData()函数封装了从socket端获取数据的过程，到此函数执行完已经将数据保存到了bPacket对象中    if (!bPacket->fillInData(fRTPInterface, packetReadWasIncomplete))    {      if (bPacket->bytesAvailable() == 0)       {      envir() << "MultiFramedRTPSource error: Hit limit when reading incoming packet over TCP. Increase \"MAX_PACKET_SIZE\"\n";      }      break;   }    if (packetReadWasIncomplete)    {      // We need additional read(s) before we can process the incoming packet:      fPacketReadInProgress = bPacket;      return;    } else     {      fPacketReadInProgress = NULL;    }        //省略关于RTP包的处理    ...    ...    ...    //fReorderingBuffer为MultiFramedRTPSource类中的对象，该对象建立了一个存储Packet数据包对象的链表    //下面的storePacket()函数即将上面获取的数据包存储在链表中    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}

 

//下面的这个函数则实现从上面函数中介绍的存储数据包链表的对象（即fReorderingBuffer）中取出数据包并调用相应函数使用它

//代码1.1

 

void MultiFramedRTPSource::doGetNextFrame1() {  while (fNeedDelivery)   {    // If we already have packet data available, then deliver it now.    Boolean packetLossPrecededThis;     //从fReorderingBuffer对象中取出一个数据包    BufferedPacket* nextPacket      = fReorderingBuffer->getNextCompletedPacket(packetLossPrecededThis);    if (nextPacket == NULL) break;    fNeedDelivery = False;    if (nextPacket->useCount() == 0)     {      // Before using the packet, check whether it has a special header      // that needs to be processed:      unsigned specialHeaderSize;      if (!processSpecialHeader(nextPacket, specialHeaderSize))      {// Something's wrong with the header; reject the packet:fReorderingBuffer->releaseUsedPacket(nextPacket);fNeedDelivery = True;break;      }      nextPacket->skip(specialHeaderSize);    }    // Check whether we're part of a multi-packet frame, and whether    // there was packet loss that would render this packet unusable:    if (fCurrentPacketBeginsFrame)     {      if (packetLossPrecededThis || fPacketLossInFragmentedFrame)       {// We didn't get all of the previous frame.// Forget any data that we used from it:fTo = fSavedTo; fMaxSize = fSavedMaxSize;fFrameSize = 0;      }      fPacketLossInFragmentedFrame = False;    } else if (packetLossPrecededThis)     {      // We're in a multi-packet frame, with preceding packet loss      fPacketLossInFragmentedFrame = True;    }    if (fPacketLossInFragmentedFrame)    {      // This packet is unusable; reject it:      fReorderingBuffer->releaseUsedPacket(nextPacket);      fNeedDelivery = True;      break;    }    // The packet is usable. Deliver all or part of it to our caller:    unsigned frameSize;    //将上面取出的数据包拷贝到fTo指针所指向的地址    nextPacket->use(fTo, fMaxSize, frameSize, fNumTruncatedBytes,    fCurPacketRTPSeqNum, fCurPacketRTPTimestamp,    fPresentationTime, fCurPacketHasBeenSynchronizedUsingRTCP,    fCurPacketMarkerBit);    fFrameSize += frameSize;    if (!nextPacket->hasUsableData())     {      // We're completely done with this packet now      fReorderingBuffer->releaseUsedPacket(nextPacket);    }    if (fCurrentPacketCompletesFrame) //如果完整的取出了一帧数据，则可调用需要该帧数据的函数去处理它     {      // We have all the data that the client wants.      if (fNumTruncatedBytes > 0)       {envir() << "MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size ("<< fSavedMaxSize << ").  "<< fNumTruncatedBytes << " bytes of trailing data will be dropped!\n";      }      // Call our own 'after getting' function, so that the downstream object can consume the data:      if (fReorderingBuffer->isEmpty())       {// Common case optimization: There are no more queued incoming packets, so this code will not get// executed again without having first returned to the event loop.  Call our 'after getting' function// directly, because there's no risk of a long chain of recursion (and thus stack overflow):afterGetting(this);  //调用函数去处理取出的数据帧       } else       {// Special case: Call our 'after getting' function via the event loop.nextTask() = envir().taskScheduler().scheduleDelayedTask(0, (TaskFunc*)FramedSource::afterGetting, this);      }    }    else         {      // This packet contained fragmented data, and does not complete      // the data that the client wants.  Keep getting data:      fTo += frameSize; fMaxSize -= frameSize;      fNeedDelivery = True;    }  }}

//下面这个函数即开始调用执行需要该帧数据的函数

void FramedSource::afterGetting(FramedSource* source) {  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);  }}

 

上面的fAfterGettingFunc为我们自己注册的函数，如果运行的是testProgs中的openRTSP实例，则该函数指向下列代码中通过调用getNextFrame（）注册的afterGettingFrame（）函数

Boolean FileSink::continuePlaying(){  if (fSource == NULL) return False;  fSource->getNextFrame(fBuffer, fBufferSize,afterGettingFrame, this,onSourceClosure, this);  return True;}

如果运行的是testProgs中的testRTSPClient中的实例，则该函数指向这里注册的afterGettingFrame（）函数

 

Boolean DummySink::continuePlaying(){  if (fSource == NULL) return False; // sanity check (should not happen)  // Request the next frame of data from our input source.  "afterGettingFrame()" will get called later, when it arrives:  fSource->getNextFrame(fReceiveBuffer, DUMMY_SINK_RECEIVE_BUFFER_SIZE,                        afterGettingFrame, this,                        onSourceClosure, this);  return True;}

 

从上面的代码中可以看到getNextFrame()函数的第一个参数为分别在各自类中定义的buffer，我们继续以openRTSP为运行程序来分析，fBuffer为FileSink类里定义的指针：unsigned char* fBuffer;

这里我们先绕一个弯，看看getNextFrame()函数里做了什么

 

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();}

 

从代码可以知道上面getNextFrame()中传入的第一个参数fBuffer指向了指针fTo，而我们在前面分析代码1.1中的void MultiFramedRTPSource::doGetNextFrame1()函数中有下面一段代码：

   //将上面取出的数据包拷贝到fTo指针所指向的地址    nextPacket->use(fTo, fMaxSize, frameSize, fNumTruncatedBytes,    fCurPacketRTPSeqNum, fCurPacketRTPTimestamp,    fPresentationTime, fCurPacketHasBeenSynchronizedUsingRTCP,    fCurPacketMarkerBit);

实际上现在应该明白了，从getNextFrame()函数中传入的第一个参数fBuffer最终存储的即是从数据包链表对象中取出的数据，并且在调用上面的use()函数后就可以使用了。
而在void MultiFramedRTPSource::doGetNextFrame1()函数中代码显示的最终调用我们注册的void FileSink::afterGettingFrame（）正好是在use()函数调用之后的afterGetting(this)中调用。我们再看看afterGettingFrame()做了什么处理：

 

void FileSink::afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes, struct timeval presentationTime, unsigned /*durationInMicroseconds*/){  FileSink* sink = (FileSink*)clientData;  sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime);}void FileSink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes, struct timeval presentationTime) {  if (numTruncatedBytes > 0)     {    envir() << "FileSink::afterGettingFrame(): The input frame data was too large for our buffer size ("    << fBufferSize << ").  "            << numTruncatedBytes << " bytes of trailing data was dropped!  Correct this by increasing the \"bufferSize\" parameter in the \"createNew()\" call to at least "            << fBufferSize + numTruncatedBytes << "\n";  }  addData(fBuffer, frameSize, presentationTime);  if (fOutFid == NULL || fflush(fOutFid) == EOF)     {    // The output file has closed.  Handle this the same way as if the    // input source had closed:    onSourceClosure(this);    stopPlaying();    return;  }  if (fPerFrameFileNameBuffer != NULL)     {    if (fOutFid != NULL) { fclose(fOutFid); fOutFid = NULL; }  }  // Then try getting the next frame:  continuePlaying();}

从上面代码可以看到调用了addData()函数将数据保存到文件中，然后继续continuePlaying()又去获取下一帧数据然后处理，直到遇到循环结束然后依次退出调用函数。最后看看addData()函数的实现即可知：

 

void FileSink::addData(unsigned char const* data, unsigned dataSize,       struct timeval presentationTime) {  if (fPerFrameFileNameBuffer != NULL)     {    // Special case: Open a new file on-the-fly for this frame    sprintf(fPerFrameFileNameBuffer, "%s-%lu.%06lu", fPerFrameFileNamePrefix,    presentationTime.tv_sec, presentationTime.tv_usec);    fOutFid = OpenOutputFile(envir(), fPerFrameFileNameBuffer);  }  // Write to our file:#ifdef TEST_LOSS  static unsigned const framesPerPacket = 10;  static unsigned const frameCount = 0;  static Boolean const packetIsLost;  if ((frameCount++)%framesPerPacket == 0)     {    packetIsLost = (our_random()%10 == 0); // simulate 10% packet loss #####  }  if (!packetIsLost)#endif  if (fOutFid != NULL && data != NULL)    {    fwrite(data, 1, dataSize, fOutFid);  }}

最后调用系统函数fwrite()实现写入文件功能。

总结：从上面的分析可知，如果要取得从RTSP服务器端接收并保存的数据帧，我们只需要定义一个类并实现如下格式两个的函数，并声明一个指针地址buffer用于指向数据帧，再在continuePlaying()函数中调用getNextFrame(buffer，...)即可。

  typedef void (afterGettingFunc)(void* clientData, unsigned frameSize,  unsigned numTruncatedBytes,  struct timeval presentationTime,  unsigned durationInMicroseconds);  typedef void (onCloseFunc)(void* clientData);

然后再在afterGettingFunc的函数中即可使用buffer。.