live555從RTSP服務器讀取數據到使用接收到的數據流程分析

進入testProgs目錄，執行./openRTSP rtsp://xxxx/test.mp4

對於RTSP協議的處理部分，可設置斷點在setupStreams函數中，並跟蹤即可進行分析。

這裏主要分析進入如下的while(1)循環中的代碼

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

 

從這裏可知，live555在客戶端處理數據實際上是單線程的程序，不斷執行SingleStep()函數中的代碼。通過查看該函數代碼裏，下面一句代碼爲重點

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

 

其中該條代碼出現了兩次，通過調試跟蹤它的執行軌跡，第一次出現調用的函數是爲了處理和RTSP服務器的通信協議的商定，而第二次出現調用的函數纔是處理真正的視頻和音頻數據。對於RTSP通信協議的分析我們暫且不討論，而直接進入第二次調用該函數的部分。

在我們的調試過程中在執行到上面的函數時就直接調用到livemedia目錄下的如下函數

 

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

//下面這個函數實現的主要功能就是從socket端讀取數據並存儲數據

 

void MultiFramedRTPSource::networkReadHandler1()
{
  BufferedPacket* bPacket = fPacketReadInProgress;
  if (bPacket == NULL)
  {
    // Normal case: Get a free BufferedPacket descriptor to hold the new network packet:
    //分配一塊新的存儲空間來存儲從socket端讀取的數據
    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()函數封裝了從socket端獲取數據的過程，到此函數執行完已經將數據保存到了bPacket對象中
    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;
    }

    //省略關於RTP包的處理
    ...
    ...
    ...
    //fReorderingBuffer爲MultiFramedRTPSource類中的對象，該對象建立了一個存儲Packet數據包對象的鏈表
    //下面的storePacket()函數即將上面獲取的數據包存儲在鏈表中
    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
}

 

//下面的這個函數則實現從上面函數中介紹的存儲數據包鏈表的對象（即fReorderingBuffer）中取出數據包並調用相應函數使用它

//代碼1.1

 

void MultiFramedRTPSource::doGetNextFrame1()
{
  while (fNeedDelivery)
  {
    // If we already have packet data available, then deliver it now.
    Boolean packetLossPrecededThis;
    //從fReorderingBuffer對象中取出一個數據包
    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;
    //將上面取出的數據包拷貝到fTo指針所指向的地址
    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幀數據，則可調用需要該幀數據的函數去處理它
     {
      // 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);  //調用函數去處理取出的數據幀
       } 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;
    }
  }
}

//下面這個函數即開始調用執行需要該幀數據的函數

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爲我們自己註冊的函數，如果運行的是testProgs中的openRTSP實例，則該函數指向下列代碼中通過調用getNextFrame（）註冊的afterGettingFrame（）函數

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

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

  return True;
}

如果運行的是testProgs中的testRTSPClient中的實例，則該函數指向這裏註冊的afterGettingFrame（）函數

 

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;
}

 

從上面的代碼中可以看到getNextFrame()函數的第一個參數爲分別在各自類中定義的buffer，我們繼續以openRTSP爲運行程序來分析，fBuffer爲FileSink類裏定義的指針：unsigned char* fBuffer;

這裏我們先繞一個彎，看看getNextFrame()函數裏做了什麼

 

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();
}

 

從代碼可以知道上面getNextFrame()中傳入的第一個參數fBuffer指向了指針fTo，而我們在前面分析代碼1.1中的void MultiFramedRTPSource::doGetNextFrame1()函數中有下面一段代碼：

//將上面取出的數據包拷貝到fTo指針所指向的地址
 nextPacket->use(fTo, fMaxSize, frameSize, fNumTruncatedBytes,
   fCurPacketRTPSeqNum, fCurPacketRTPTimestamp,
   fPresentationTime, fCurPacketHasBeenSynchronizedUsingRTCP,
   fCurPacketMarkerBit);

實際上現在應該明白了，從getNextFrame()函數中傳入的第一個參數fBuffer最終存儲的即是從數據包鏈表對象中取出的數據，並且在調用上面的use()函數後就可以使用了。
而在void MultiFramedRTPSource::doGetNextFrame1()函數中代碼顯示的最終調用我們註冊的void FileSink::afterGettingFrame（）正好是在use()函數調用之後的afterGetting(this)中調用。我們再看看afterGettingFrame()做了什麼處理：

 

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();
}

從上面代碼可以看到調用了addData()函數將數據保存到文件中，然後繼續continuePlaying()又去獲取下一幀數據然後處理，直到遇到循環結束然後依次退出調用函數。最後看看addData()函數的實現即可知：

 

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);
  }
}

最後調用系統函數fwrite()實現寫入文件功能。

總結：從上面的分析可知，如果要取得從RTSP服務器端接收並保存的數據幀，我們只需要定義一個類並實現如下格式兩個的函數，並聲明一個指針地址buffer用於指向數據幀，再在continuePlaying()函數中調用getNextFrame(buffer，...)即可。

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

然後再在afterGettingFunc的函數中即可使用buffer。.