live555客户端OpenRtsp流程

 

以下是我个人总结的（桂堂东的博客给了我很大的帮助。一会把他那篇转出来）：

程序从rtsp_player_task这个线程开始进入进入到live555的客户端。

1：char *argv[5]={"openRTSP", "-b", "80000", "-t", ""};这是输入参数。b代表FileSink的大小具体值为80000. T代表用tcp   run_live_rtsp(int argc, char **argv) 这个函数来处理这些参数。

2：在run_live_rtsp(int argc, char **argv) 这个函数里接着:

ourClient = createClient(*env, verbosityLevel, progName);//来创建一个客户端实例。char* optionsResponse = getOptionsResponse(ourClient, url, username, password);// 来发送接受并判断options。char* sdpDescription= getSDPDescriptionFromURL(ourClient, url, username, password, proxyServerName, proxyServerPortNum,desiredPortNum);//这个函数用来用来发送接受description，并得到sdp： 

v=0o=- 1266888546188420 1 IN IP4 10.0.4.152s=MPEG Transport Stream, streamed by the LIVE555 Media Serveri=question.tst=0 0a=tool:LIVE555 Streaming Media v2009.07.09a=type:broadcasta=control:*a=range:npt=0-182.512a=x-qt-text-nam:MPEG Transport Stream, streamed by the LIVE555 Media Servera=x-qt-text-inf:question.tsm=video 0 RTP/AVP 33c=IN IP4 0.0.0.0a=control:track1 

 

//根据sdp创建会话session = MediaSession::createNew(*env, sdpDescription);MediaSession* newSession = new MediaSession(env);//创建会话。newSession->initializeWithSDP(sdpDescription);//初始化变量 最主要是依据m=。。。。来创建子会话和选用rtp协议subsession->initiate(simpleRTPoffsetArg);//创建接收区的一些东西进入initiate仔细看。//首先是fRTPSocket = new Groupsock(env(), tempAddr, rtpPortNum, 255);//设置rtpsocket。//调用到OutputSocket-------》Socket(env, port)------》setupDatagramSocke在这个函数里面int newSocket = socket(AF_INET, SOCK_DGRAM, 0);//创建newsocket。。port.num() =0 ReceivingInterfaceAddr==0 INADDR_ANY==0adddddddr=0//所以 if (port.num() != 0 || ReceivingInterfaceAddr != INADDR_ANY)//就没进去。。。//在 if (!socketJoinGroup(env, socketNum(), groupAddr.s_addr))//里if (!IsMulticastAddress(groupAddress)) return True;//直接返回然后再ourSourceAddressForMulticast//这个函数里调用setupDatagramSocke//t函数，//里面由于有了port所以会执行bind。//回到ourSourceAddressForMulticast//里 if (!socketJoinGroup(env, sock, testAddr.s_addr))//就可以加入成功了//。writeSocket和 readSocket是通过写和读hostIdTest来获取本地ip。。。//这个fRTPSocket = new Groupsock(env(), tempAddr, rtpPortNum, 255);//其实就是申请一个socket放在那，然后再申请一个，绑定后writeSocket和 readSocket组播得到本机ip然后撤销socket....//getSourcePort(env(), fRTPSocket->socketNum(), clientPort)随即得到一个rtp端口号然后把fRTPSocket = new Groupsock第一次创建的那个socket和这个rtp端口进行绑定。 

 

 

 

//而frtcpsocket没有加入到组播地址里。接着依据fCodecName来创建fRTPSource和fReadSource。//由于我的ts流是MP2T的所以if (strcmp(fCodecName, "MP2T") == 0){ // MPEG-2 Transport Stream printf("keke/n"); fRTPSource = SimpleRTPSource::createNew(env(),fRTPSocket, fRTPPayloadFormat,fRTPTimestampFrequency, "video/MP2T",0, False); fReadSource = MPEG2TransportStreamFramer::createNew(env(), fRTPSource);printf("defr/n"); // this sets "durationInMicroseconds" correctly, based on the PCR values } //fRTPSource创造媒体流名字并加入列表，而fReadSource有点像filter其输入端像是fRTPSource。fRTCPInstance = RTCPInstance::createNew(env(), fRTCPSocket, totSessionBandwidth, (unsigned char const*) fParent.CNAME(), NULL /* we're a client */, fRTPSource);//创造fRTCPInstance进行rtp的控制。 

 

 

//现在进入setupStreams();一步一步调用到setupMediaSubsession它主要完成Sending request: SETUP rtsp://10.0.4.152/question.ts/track1 RTSP/1.0CSeq: 3Transport: RTP/AVP;unicast;client_port=32770-32771User-Agent: openRTSP (LIVE555 Streaming Media v2006.11.15)本地的端口号和获取server的端口号Received SETUP response: RTSP/1.0 200 OKCSeq: 3Date: Mon, Mar 01 2010 07:56:37 GMTTransport: RTP/AVP;unicast;destination=10.0.4.155;source=10.0.4.152;client_port=32770-32771;server_port=6970-6971Session: 16 

 

 

fileSink = FileSink::createNew(*env, outFileName, fileSinkBufferSize, oneFilePerFrame); if (strcmp(subsession->mediumName(), "video") == 0){ printf("fileSink->set_media_type(1);/n"); fileSink->set_media_type(1);}//创建sink文件并设置文件类型。 subsession->sink->startPlaying(*(subsession->readSource()),subsessionAfterPlaying, subsession);//开始取数据 

 

startPlayingStreams(-1);告诉server可以发数据了

 

//现在进入不断循环取数据的阶段：env->taskScheduler().doEventLoop()//------->BasicTaskScheduler::SingleStep//它首先接受描述符fd_set readSet = fReadSet; //而fReadSet在之前就已经创建了并且通过turnOnBackgroundReadHandling(int socketNum,BackgroundHandlerProc* handlerProc,void* clientData)// 这个函数把每个套接口加入进去并组织好了队列。接下来就是不断循环看哪个套接口可读呢。//然后调用MultiFramedRTPSource:: networkReadHandler//这个函数来读取数据。bPacket->fillInData(source->fRTPInterface)//取数据//最后调用到Boolean RTPInterface::handleRead(unsigned char* buffer,unsigned bufferMaxSize,unsigned& bytesRead,struct sockaddr_in& fromAddress) //由于用的是传输层用的是udp所以fNextTCPReadStreamSocketNum<0//调用readSuccess = fGS->handleRead(buffer, bufferMaxSize, bytesRead, fromAddress);//调用到Boolean Groupsock::handleRead(unsigned char* buffer, unsigned bufferMaxSize,unsigned& bytesRead,struct sockaddr_in& fromAddress)//接着调用到int readSocket(UsageEnvironment& env,int socket, unsigned char* buffer, unsigned bufferSize,struct sockaddr_in& fromAddress,struct timeval* timeout) //里面bytesRead = recvfrom(socket, (char*)buffer, bufferSize, 0,(struct sockaddr*)&fromAddress,&addressSize);//读取数据statsIncoming.countPacket(numBytes); statsGroupIncoming.countPacket(numBytes);//这两个函数统计读入的字节数。//fillInData这个函数到此就执行完。//接下来的工作是分析包头直到The rest of the packet is the usable data. Record and save it: storePacket(bPacket)//这个函数是把bPacket组织成队列形式，//接下来调用到MultiFramedRTPSource::doGetNextFrame1()//别被他名字骗了，其实他就是去取我们刚刚得到的那包数据BufferedPacket* nextPacket = fReorderingBuffer->getNextCompletedPacket(packetLossPrecededThis);//然后判断是不是我们希望的。然后调用ramedSource::afterGetting//，接着调用FileSink::afterGettingFrame//接着FileSink::afterGettingFrame1//然后addData(unsigned char* data, unsigned dataSize,struct timeval presentationTime)//这时调用notify_rtp_frame(get_media_type(), data, dataSize,presentationTime);//这个函数告诉dma有数据来了 continuePlaying();//用从头开始循环

 

原文：http://blog.csdn.net/wangyi84971224/archive/2010/04/07/5457986.aspx