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"
";
      }
      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!
";
      }
      // 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!
";
    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 << "
";
  }
  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。.