(转)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。
- 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。.
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