live555学习笔记8－RTSPClient分析

八 RTSPClient分析

有RTSPServer，当然就要有RTSPClient。
如果按照Server端的架构，想一下Client端各部分的组成可能是这样：
因为要连接RTSP server，所以RTSPClient要有TCP socket。当获取到server端的DESCRIBE后，应建立一个对应于ServerMediaSession的ClientMediaSession。对应每个Track，ClientMediaSession中应建立ClientMediaSubsession。当建立RTP Session时，应分别为所拥有的Track发送SETUP请求连接，在获取回应后，分别为所有的track建立RTP socket，然后请求PLAY，然后开始传输数据。事实是这样吗？只能分析代码了。

testProgs中的OpenRTSP是典型的RTSPClient示例，所以分析它吧。
main()函数在playCommon.cpp文件中。main()的流程比较简单，跟服务端差别不大：建立任务计划对象－－建立环境对象－－处理用户输入的参数(RTSP地址)－－创建RTSPClient实例－－发出第一个RTSP请求（可能是OPTIONS也可能是DESCRIBE）－－进入Loop。

RTSP的tcp连接是在发送第一个RTSP请求时才建立的，在RTSPClient的那几个发请求的函数sendXXXXXXCommand()中最终都调用sendRequest()，sendRequest()中会跟据情况建立起TCP连接。在建立连接时马上向任务计划中加入处理从这个TCP接收数据的socket handler：RTSPClient::incomingDataHandler()。
下面就是发送RTSP请求，OPTIONS就不必看了，从请求DESCRIBE开始：

void getSDPDescription(RTSPClient::responseHandler* afterFunc){ourRTSPClient->sendDescribeCommand(afterFunc, ourAuthenticator);}unsigned RTSPClient::sendDescribeCommand(responseHandler* responseHandler,Authenticator* authenticator){if (authenticator != NULL)fCurrentAuthenticator = *authenticator;return sendRequest(new RequestRecord(++fCSeq, "DESCRIBE", responseHandler));}

参数responseHandler是调用者提供的回调函数，用于在处理完请求的回应后再调用之。并且在这个回调函数中会发出下一个请求－－所有的请求都是这样依次发出的。使用回调函数的原因主要是因为socket的发送与接收不是同步进行的。类RequestRecord就代表一个请求,它不但保存了RTSP请求相关的信息，而且保存了请求完成后的回调函数－－就是responseHandler。有些请求发出时还没建立tcp连接，不能立即发送，则加入fRequestsAwaitingConnection队列；有些发出后要等待Server端的回应，就加入fRequestsAwaitingResponse队列，当收到回应后再从队列中把它取出。
由于RTSPClient::sendRequest()太复杂，就不列其代码了，其无非是建立起RTSP请求字符串然后用TCP socket发送之。

现在看一下收到DESCRIBE的回应后如何处理它。理论上是跟据媒体信息建立起MediaSession了，看看是不是这样：

void continueAfterDESCRIBE(RTSPClient*, int resultCode, char* resultString){char* sdpDescription = resultString;//跟据SDP创建MediaSession。// Create a media session object from this SDP description:session = MediaSession::createNew(*env, sdpDescription);delete[] sdpDescription;// Then, setup the "RTPSource"s for the session:MediaSubsessionIterator iter(*session);MediaSubsession *subsession;Boolean madeProgress = False;char const* singleMediumToTest = singleMedium;//循环所有的MediaSubsession，为每个设置其RTPSource的参数while ((subsession = iter.next()) != NULL) {//初始化subsession，在其中会建立RTP/RTCP socket以及RTPSource。if (subsession->initiate(simpleRTPoffsetArg)) {madeProgress = True;if (subsession->rtpSource() != NULL) {// Because we're saving the incoming data, rather than playing// it in real time, allow an especially large time threshold// (1 second) for reordering misordered incoming packets:unsigned const thresh = 1000000; // 1 secondsubsession->rtpSource()->setPacketReorderingThresholdTime(thresh);// Set the RTP source's OS socket buffer size as appropriate - either if we were explicitly asked (using -B),// or if the desired FileSink buffer size happens to be larger than the current OS socket buffer size.// (The latter case is a heuristic, on the assumption that if the user asked for a large FileSink buffer size,// then the input data rate may be large enough to justify increasing the OS socket buffer size also.)int socketNum = subsession->rtpSource()->RTPgs()->socketNum();unsigned curBufferSize = getReceiveBufferSize(*env,socketNum);if (socketInputBufferSize > 0　|| fileSinkBufferSize > curBufferSize) {unsigned newBufferSize =　socketInputBufferSize > 0 ?　socketInputBufferSize :fileSinkBufferSize;newBufferSize = setReceiveBufferTo(*env, socketNum,newBufferSize);if (socketInputBufferSize > 0) { // The user explicitly asked for the new socket buffer size; announce it:*env<< "Changed socket receive buffer size for the \""<< subsession->mediumName() << "/"<< subsession->codecName()<< "\" subsession from " << curBufferSize<< " to " << newBufferSize << " bytes\n";}}}}}if (!madeProgress)shutdown();// Perform additional 'setup' on each subsession, before playing them://下一步就是发送SETUP请求了。需要为每个Track分别发送一次。setupStreams();}

此函数被删掉很多枝叶，所以发现与原版不同请不要惊掉大牙。
的确在DESCRIBE回应后建立起了MediaSession，而且我们发现Client端的MediaSession不叫ClientMediaSesson，SubSession亦不是。我现在很想看看MediaSession与MediaSubsession的建立过程:

MediaSession* MediaSession::createNew(UsageEnvironment& env,char const* sdpDescription){MediaSession* newSession = new MediaSession(env);if (newSession != NULL) {if (!newSession->initializeWithSDP(sdpDescription)) {delete newSession;return NULL;}}return newSession;}

我可以告诉你，MediaSession的构造函数没什么可看的，那么就来看initializeWithSDP()：
内容太多，不必看了，我大体说说吧：就是处理SDP，跟据每一行来初始化一些变量。当遇到"m="行时，就建立一个MediaSubsession，然后再处理这一行之下，下一个"m="行之上的行们，用这些参数初始化MediaSubsession的变量。循环往复，直到尽头。然而这其中并没有建立RTP socket。我们发现在continueAfterDESCRIBE()中，创建MediaSession之后又调用了subsession->initiate(simpleRTPoffsetArg)，那么socket是不是在它里面创建的呢？look:

Boolean MediaSubsession::initiate(int useSpecialRTPoffset){if (fReadSource != NULL)return True; // has already been initiateddo {if (fCodecName == NULL) {env().setResultMsg("Codec is unspecified");break;}//创建RTP/RTCP sockets// Create RTP and RTCP 'Groupsocks' on which to receive incoming data.// (Groupsocks will work even for unicast addresses)struct in_addr tempAddr;tempAddr.s_addr = connectionEndpointAddress();// This could get changed later, as a result of a RTSP "SETUP"if (fClientPortNum != 0) {//当server端指定了建议的client端口// The sockets' port numbers were specified for us.  Use these:fClientPortNum = fClientPortNum & ~1; // evenif (isSSM()) {fRTPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr,fClientPortNum);} else {fRTPSocket = new Groupsock(env(), tempAddr, fClientPortNum,255);}if (fRTPSocket == NULL) {env().setResultMsg("Failed to create RTP socket");break;}// Set our RTCP port to be the RTP port +1portNumBits const rtcpPortNum = fClientPortNum | 1;if (isSSM()) {fRTCPSocket = new Groupsock(env(), tempAddr, fSourceFilterAddr,rtcpPortNum);} else {fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum, 255);}if (fRTCPSocket == NULL) {char tmpBuf[100];sprintf(tmpBuf, "Failed to create RTCP socket (port %d)",rtcpPortNum);env().setResultMsg(tmpBuf);break;}} else {//Server端没有指定client端口，我们自己找一个。之所以做的这样复杂，是为了能找到连续的两个端口//RTP/RTCP的端口号不是要连续吗？还记得不？// Port numbers were not specified in advance, so we use ephemeral port numbers.// Create sockets until we get a port-number pair (even: RTP; even+1: RTCP).// We need to make sure that we don't keep trying to use the same bad port numbers over and over again.// so we store bad sockets in a table, and delete them all when we're done.HashTable* socketHashTable = HashTable::create(ONE_WORD_HASH_KEYS);if (socketHashTable == NULL)break;Boolean success = False;NoReuse dummy; // ensures that our new ephemeral port number won't be one that's already in usewhile (1) {// Create a new socket:if (isSSM()) {fRTPSocket = new Groupsock(env(), tempAddr,fSourceFilterAddr, 0);} else {fRTPSocket = new Groupsock(env(), tempAddr, 0, 255);}if (fRTPSocket == NULL) {env().setResultMsg("MediaSession::initiate(): unable to create RTP and RTCP sockets");break;}// Get the client port number, and check whether it's even (for RTP):Port clientPort(0);if (!getSourcePort(env(), fRTPSocket->socketNum(),clientPort)) {break;}fClientPortNum = ntohs(clientPort.num());if ((fClientPortNum & 1) != 0) { // it's odd// Record this socket in our table, and keep trying:unsigned key = (unsigned) fClientPortNum;Groupsock* existing = (Groupsock*) socketHashTable->Add((char const*) key, fRTPSocket);delete existing; // in case it wasn't NULLcontinue;}// Make sure we can use the next (i.e., odd) port number, for RTCP:portNumBits rtcpPortNum = fClientPortNum | 1;if (isSSM()) {fRTCPSocket = new Groupsock(env(), tempAddr,fSourceFilterAddr, rtcpPortNum);} else {fRTCPSocket = new Groupsock(env(), tempAddr, rtcpPortNum,255);}if (fRTCPSocket != NULL && fRTCPSocket->socketNum() >= 0) {// Success! Use these two sockets.success = True;break;} else {// We couldn't create the RTCP socket (perhaps that port number's already in use elsewhere?).delete fRTCPSocket;// Record the first socket in our table, and keep trying:unsigned key = (unsigned) fClientPortNum;Groupsock* existing = (Groupsock*) socketHashTable->Add((char const*) key, fRTPSocket);delete existing; // in case it wasn't NULLcontinue;}}// Clean up the socket hash table (and contents):Groupsock* oldGS;while ((oldGS = (Groupsock*) socketHashTable->RemoveNext()) != NULL) {delete oldGS;}delete socketHashTable;if (!success)break; // a fatal error occurred trying to create the RTP and RTCP sockets; we can't continue}// Try to use a big receive buffer for RTP - at least 0.1 second of// specified bandwidth and at least 50 KBunsigned rtpBufSize = fBandwidth * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytesif (rtpBufSize < 50 * 1024)rtpBufSize = 50 * 1024;increaseReceiveBufferTo(env(), fRTPSocket->socketNum(), rtpBufSize);// ASSERT: fRTPSocket != NULL && fRTCPSocket != NULLif (isSSM()) {// Special case for RTCP SSM: Send RTCP packets back to the source via unicast:fRTCPSocket->changeDestinationParameters(fSourceFilterAddr, 0, ~0);}//创建RTPSource的地方// Create "fRTPSource" and "fReadSource":if (!createSourceObjects(useSpecialRTPoffset))break;if (fReadSource == NULL) {env().setResultMsg("Failed to create read source");break;}// Finally, create our RTCP instance. (It starts running automatically)if (fRTPSource != NULL) {// If bandwidth is specified, use it and add 5% for RTCP overhead.// Otherwise make a guess at 500 kbps.unsigned totSessionBandwidth =fBandwidth ? fBandwidth + fBandwidth / 20 : 500;fRTCPInstance = RTCPInstance::createNew(env(), fRTCPSocket,totSessionBandwidth, (unsigned char const*) fParent.CNAME(),NULL /* we're a client */, fRTPSource);if (fRTCPInstance == NULL) {env().setResultMsg("Failed to create RTCP instance");break;}}return True;} while (0);//失败时执行到这里delete fRTPSocket;fRTPSocket = NULL;delete fRTCPSocket;fRTCPSocket = NULL;Medium::close(fRTCPInstance);fRTCPInstance = NULL;Medium::close(fReadSource);fReadSource = fRTPSource = NULL;fClientPortNum = 0;return False;}

是的，在其中创建了RTP/RTCP socket并创建了RTPSource，创建RTPSource在函数createSourceObjects()中，看一下：

Boolean MediaSubsession::createSourceObjects(int useSpecialRTPoffset){do {// First, check "fProtocolName"if (strcmp(fProtocolName, "UDP") == 0) {// A UDP-packetized stream (*not* a RTP stream)fReadSource = BasicUDPSource::createNew(env(), fRTPSocket);fRTPSource = NULL; // Note!if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport StreamfReadSource = MPEG2TransportStreamFramer::createNew(env(),fReadSource);// this sets "durationInMicroseconds" correctly, based on the PCR values}} else {// Check "fCodecName" against the set of codecs that we support,// and create our RTP source accordingly// (Later make this code more efficient, as this set grows #####)// (Also, add more fmts that can be implemented by SimpleRTPSource#####)Boolean createSimpleRTPSource = False; // by default; can be changed belowBoolean doNormalMBitRule = False; // default behavior if "createSimpleRTPSource" is Trueif (strcmp(fCodecName, "QCELP") == 0) { // QCELP audiofReadSource = QCELPAudioRTPSource::createNew(env(), fRTPSocket,fRTPSource, fRTPPayloadFormat, fRTPTimestampFrequency);// Note that fReadSource will differ from fRTPSource in this case} else if (strcmp(fCodecName, "AMR") == 0) { // AMR audio (narrowband)fReadSource = AMRAudioRTPSource::createNew(env(), fRTPSocket,fRTPSource, fRTPPayloadFormat, 0 /*isWideband*/,fNumChannels, fOctetalign, fInterleaving,fRobustsorting, fCRC);// Note that fReadSource will differ from fRTPSource in this case} else if (strcmp(fCodecName, "AMR-WB") == 0) { // AMR audio (wideband)fReadSource = AMRAudioRTPSource::createNew(env(), fRTPSocket,fRTPSource, fRTPPayloadFormat, 1 /*isWideband*/,fNumChannels, fOctetalign, fInterleaving,fRobustsorting, fCRC);// Note that fReadSource will differ from fRTPSource in this case} else if (strcmp(fCodecName, "MPA") == 0) { // MPEG-1 or 2 audiofReadSource = fRTPSource = MPEG1or2AudioRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,fRTPTimestampFrequency);} else if (strcmp(fCodecName, "MPA-ROBUST") == 0) { // robust MP3 audiofRTPSource = MP3ADURTPSource::createNew(env(), fRTPSocket,fRTPPayloadFormat, fRTPTimestampFrequency);if (fRTPSource == NULL)break;// Add a filter that deinterleaves the ADUs after depacketizing them:MP3ADUdeinterleaver* deinterleaver = MP3ADUdeinterleaver::createNew(env(), fRTPSource);if (deinterleaver == NULL)break;// Add another filter that converts these ADUs to MP3 frames:fReadSource = MP3FromADUSource::createNew(env(), deinterleaver);} else if (strcmp(fCodecName, "X-MP3-DRAFT-00") == 0) {// a non-standard variant of "MPA-ROBUST" used by RealNetworks// (one 'ADU'ized MP3 frame per packet; no headers)fRTPSource = SimpleRTPSource::createNew(env(), fRTPSocket,fRTPPayloadFormat, fRTPTimestampFrequency,"audio/MPA-ROBUST" /*hack*/);if (fRTPSource == NULL)break;// Add a filter that converts these ADUs to MP3 frames:fReadSource = MP3FromADUSource::createNew(env(), fRTPSource,False /*no ADU header*/);} else if (strcmp(fCodecName, "MP4A-LATM") == 0) { // MPEG-4 LATM audiofReadSource = fRTPSource = MPEG4LATMAudioRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,fRTPTimestampFrequency);} else if (strcmp(fCodecName, "AC3") == 0|| strcmp(fCodecName, "EAC3") == 0) { // AC3 audiofReadSource = fRTPSource = AC3AudioRTPSource::createNew(env(),fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency);} else if (strcmp(fCodecName, "MP4V-ES") == 0) { // MPEG-4 Elem Str vidfReadSource = fRTPSource = MPEG4ESVideoRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,fRTPTimestampFrequency);} else if (strcmp(fCodecName, "MPEG4-GENERIC") == 0) {fReadSource = fRTPSource = MPEG4GenericRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,fRTPTimestampFrequency, fMediumName, fMode, fSizelength,fIndexlength, fIndexdeltalength);} else if (strcmp(fCodecName, "MPV") == 0) { // MPEG-1 or 2 videofReadSource = fRTPSource = MPEG1or2VideoRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,fRTPTimestampFrequency);} else if (strcmp(fCodecName, "MP2T") == 0) { // MPEG-2 Transport StreamfRTPSource = SimpleRTPSource::createNew(env(), fRTPSocket,fRTPPayloadFormat, fRTPTimestampFrequency, "video/MP2T",0, False);fReadSource = MPEG2TransportStreamFramer::createNew(env(),fRTPSource);// this sets "durationInMicroseconds" correctly, based on the PCR values} else if (strcmp(fCodecName, "H261") == 0) { // H.261fReadSource = fRTPSource = H261VideoRTPSource::createNew(env(),fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency);} else if (strcmp(fCodecName, "H263-1998") == 0|| strcmp(fCodecName, "H263-2000") == 0) { // H.263+fReadSource = fRTPSource = H263plusVideoRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,fRTPTimestampFrequency);} else if (strcmp(fCodecName, "H264") == 0) {fReadSource = fRTPSource = H264VideoRTPSource::createNew(env(),fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency);} else if (strcmp(fCodecName, "DV") == 0) {fReadSource = fRTPSource = DVVideoRTPSource::createNew(env(),fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency);} else if (strcmp(fCodecName, "JPEG") == 0) { // motion JPEGfReadSource = fRTPSource = JPEGVideoRTPSource::createNew(env(),fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency,videoWidth(), videoHeight());} else if (strcmp(fCodecName, "X-QT") == 0|| strcmp(fCodecName, "X-QUICKTIME") == 0) {// Generic QuickTime streams, as defined in// <http://developer.apple.com/quicktime/icefloe/dispatch026.html>char* mimeType = new char[strlen(mediumName())+ strlen(codecName()) + 2];sprintf(mimeType, "%s/%s", mediumName(), codecName());fReadSource = fRTPSource = QuickTimeGenericRTPSource::createNew(env(), fRTPSocket, fRTPPayloadFormat,fRTPTimestampFrequency, mimeType);delete[] mimeType;} else if (strcmp(fCodecName, "PCMU") == 0 // PCM u-law audio|| strcmp(fCodecName, "GSM") == 0 // GSM audio|| strcmp(fCodecName, "DVI4") == 0 // DVI4 (IMA ADPCM) audio|| strcmp(fCodecName, "PCMA") == 0 // PCM a-law audio|| strcmp(fCodecName, "MP1S") == 0 // MPEG-1 System Stream|| strcmp(fCodecName, "MP2P") == 0 // MPEG-2 Program Stream|| strcmp(fCodecName, "L8") == 0 // 8-bit linear audio|| strcmp(fCodecName, "L16") == 0 // 16-bit linear audio|| strcmp(fCodecName, "L20") == 0 // 20-bit linear audio (RFC 3190)|| strcmp(fCodecName, "L24") == 0 // 24-bit linear audio (RFC 3190)|| strcmp(fCodecName, "G726-16") == 0 // G.726, 16 kbps|| strcmp(fCodecName, "G726-24") == 0 // G.726, 24 kbps|| strcmp(fCodecName, "G726-32") == 0 // G.726, 32 kbps|| strcmp(fCodecName, "G726-40") == 0 // G.726, 40 kbps|| strcmp(fCodecName, "SPEEX") == 0 // SPEEX audio|| strcmp(fCodecName, "T140") == 0 // T.140 text (RFC 4103)|| strcmp(fCodecName, "DAT12") == 0 // 12-bit nonlinear audio (RFC 3190)) {createSimpleRTPSource = True;useSpecialRTPoffset = 0;} else if (useSpecialRTPoffset >= 0) {// We don't know this RTP payload format, but try to receive// it using a 'SimpleRTPSource' with the specified header offset:createSimpleRTPSource = True;} else {env().setResultMsg("RTP payload format unknown or not supported");break;}if (createSimpleRTPSource) {char* mimeType = new char[strlen(mediumName())+ strlen(codecName()) + 2];sprintf(mimeType, "%s/%s", mediumName(), codecName());fReadSource = fRTPSource = SimpleRTPSource::createNew(env(),fRTPSocket, fRTPPayloadFormat, fRTPTimestampFrequency,mimeType, (unsigned) useSpecialRTPoffset,doNormalMBitRule);delete[] mimeType;}}return True;} while (0);return False; // an error occurred}

可以看到，这个函数里主要是跟据前面分析出的媒体和传输信息建立合适的Source。

socket建立了，Source也创建了，下一步应该是连接Sink，形成一个流。到此为止还未看到Sink的影子，应该是在下一步SETUP中建立，我们看到在continueAfterDESCRIBE()的最后调用了setupStreams（），那么就来探索一下setupStreams():

void setupStreams(){static MediaSubsessionIterator* setupIter = NULL;if (setupIter == NULL)setupIter = new MediaSubsessionIterator(*session);//每次调用此函数只为一个Subsession发出SETUP请求。while ((subsession = setupIter->next()) != NULL) {// We have another subsession left to set up:if (subsession->clientPortNum() == 0)continue; // port # was not set//为一个Subsession发送SETUP请求。请求处理完成时调用continueAfterSETUP()，//continueAfterSETUP()又调用了setupStreams()，在此函数中为下一个SubSession发送SETUP请求。

//直到处理完所有的SubSessionsetupSubsession(subsession, streamUsingTCP, continueAfterSETUP);return;}//执行到这里时，已循环完所有的SubSession了// We're done setting up subsessions.delete setupIter;if (!madeProgress)shutdown();//创建输出文件，看来是在这里创建Sink了。创建sink后，就开始播放它。这个播放应该只是把socket的handler加入到//计划任务中，而没有数据的接收或发送。只有等到发出PLAY请求后才有数据的收发。// Create output files:if (createReceivers) {if (outputQuickTimeFile) {// Create a "QuickTimeFileSink", to write to 'stdout':qtOut = QuickTimeFileSink::createNew(*env, *session, "stdout",fileSinkBufferSize, movieWidth, movieHeight, movieFPS,packetLossCompensate, syncStreams, generateHintTracks,generateMP4Format);if (qtOut == NULL) {*env << "Failed to create QuickTime file sink for stdout: "<< env->getResultMsg();shutdown();}qtOut->startPlaying(sessionAfterPlaying, NULL);} else if (outputAVIFile) {// Create an "AVIFileSink", to write to 'stdout':aviOut = AVIFileSink::createNew(*env, *session, "stdout",fileSinkBufferSize, movieWidth, movieHeight, movieFPS,packetLossCompensate);if (aviOut == NULL) {*env << "Failed to create AVI file sink for stdout: "<< env->getResultMsg();shutdown();}aviOut->startPlaying(sessionAfterPlaying, NULL);} else {// Create and start "FileSink"s for each subsession:madeProgress = False;MediaSubsessionIterator iter(*session);while ((subsession = iter.next()) != NULL) {if (subsession->readSource() == NULL)continue; // was not initiated// Create an output file for each desired stream:char outFileName[1000];if (singleMedium == NULL) {// Output file name is//     "<filename-prefix><medium_name>-<codec_name>-<counter>"static unsigned streamCounter = 0;snprintf(outFileName, sizeof outFileName, "%s%s-%s-%d",fileNamePrefix, subsession->mediumName(),subsession->codecName(), ++streamCounter);} else {sprintf(outFileName, "stdout");}FileSink* fileSink;if (strcmp(subsession->mediumName(), "audio") == 0&& (strcmp(subsession->codecName(), "AMR") == 0|| strcmp(subsession->codecName(), "AMR-WB")== 0)) {// For AMR audio streams, we use a special sink that inserts AMR frame hdrs:fileSink = AMRAudioFileSink::createNew(*env, outFileName,fileSinkBufferSize, oneFilePerFrame);} else if (strcmp(subsession->mediumName(), "video") == 0&& (strcmp(subsession->codecName(), "H264") == 0)) {// For H.264 video stream, we use a special sink that insert start_codes:fileSink = H264VideoFileSink::createNew(*env, outFileName,subsession->fmtp_spropparametersets(),fileSinkBufferSize, oneFilePerFrame);} else {// Normal case:fileSink = FileSink::createNew(*env, outFileName,fileSinkBufferSize, oneFilePerFrame);}subsession->sink = fileSink;if (subsession->sink == NULL) {*env << "Failed to create FileSink for \"" << outFileName<< "\": " << env->getResultMsg() << "\n";} else {if (singleMedium == NULL) {*env << "Created output file: \"" << outFileName<< "\"\n";} else {*env << "Outputting data from the \""<< subsession->mediumName() << "/"<< subsession->codecName()<< "\" subsession to 'stdout'\n";}if (strcmp(subsession->mediumName(), "video") == 0&& strcmp(subsession->codecName(), "MP4V-ES") == 0 &&subsession->fmtp_config() != NULL) {// For MPEG-4 video RTP streams, the 'config' information// from the SDP description contains useful VOL etc. headers.// Insert this data at the front of the output file:unsignedconfigLen;unsigned char* configData= parseGeneralConfigStr(subsession->fmtp_config(), configLen);struct timeval timeNow;gettimeofday(&timeNow, NULL);fileSink->addData(configData, configLen, timeNow);delete[] configData;}//开始传输subsession->sink->startPlaying(*(subsession->readSource()),subsessionAfterPlaying, subsession);// Also set a handler to be called if a RTCP "BYE" arrives// for this subsession:if (subsession->rtcpInstance() != NULL) {subsession->rtcpInstance()->setByeHandler(subsessionByeHandler, subsession);}madeProgress = True;}}if (!madeProgress)shutdown();}}// Finally, start playing each subsession, to start the data flow:if (duration == 0) {if (scale > 0)duration = session->playEndTime() - initialSeekTime; // use SDP end timeelse if (scale < 0)duration = initialSeekTime;}if (duration < 0)duration = 0.0;endTime = initialSeekTime;if (scale > 0) {if (duration <= 0)endTime = -1.0f;elseendTime = initialSeekTime + duration;} else {endTime = initialSeekTime - duration;if (endTime < 0)endTime = 0.0f;}//发送PLAY请求，之后才能从Server端接收数据startPlayingSession(session, initialSeekTime, endTime, scale,continueAfterPLAY);}

仔细看看注释，应很容易了解此函数。