参照openRTSP写的一个RTSP client 加了一些注解

 

#include "liveMedia.hh" #include "BasicUsageEnvironment.hh" #include "GroupsockHelper.hh" UsageEnvironment* env; portNumBits tunnelOverHTTPPortNum = 0; const char * url="rtsp://127.0.0.1:1935/vod/Extremists.m4v"; #if defined(__WIN32__) || defined(_WIN32) #define snprintf _snprintf #endif int main(int argc,const char ** argv) { //创建BasicTaskScheduler对象 TaskScheduler* scheduler = BasicTaskScheduler::createNew(); //创建BisicUsageEnvironment对象 env = BasicUsageEnvironment::createNew(*scheduler); //创建RTSPClient对象 RTSPClient * rtspClient= RTSPClient::createNew(*env); //由RTSPClient对象向服务器发送OPTION消息并接受回应 char* optionsResponse=rtspClient->sendOptionsCmd(url); delete [] optionsResponse; //产生SDPDescription字符串（由RTSPClient对象向服务器发送DESCRIBE消息并接受回应，根据回应的信息产生SDPDescription字符串，其中包括视音频数据的协议和解码器类型） char* sdpDescription =rtspClient->describeURL(url); //创建MediaSession对象（根据SDPDescription在MediaSession中创建和初始化MediaSubSession子会话对象） MediaSession* session = MediaSession::createNew(*env, sdpDescription); delete[] sdpDescription; /* while循环中配置所有子会话对象 */ MediaSubsessionIterator iter(*session); MediaSubsession *subsession; while ((subsession = iter.next()) != NULL) { // Creates a "RTPSource" for this subsession. (Has no effect if it's // already been created.) Returns True iff this succeeds. if (!subsession->initiate()) { *env << "Unable to create receiver for /"" << subsession->mediumName() << "/" << subsession->codecName() << "/" subsession: " << env->getResultMsg() << "/n"; } else { *env << "Created receiver for /"" << subsession->mediumName() << "/" << subsession->codecName() << "/" subsession (client ports " << subsession->clientPortNum() << "-" << subsession->clientPortNum()+1 << ")/n"; 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 second subsession->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); unsigned newBufferSize = setReceiveBufferTo(*env, socketNum, 100000); } } } //由RTSPClient对象向服务器发送SETUP消息并接受回应 iter.reset(); while ((subsession = iter.next()) != NULL) { if (subsession->clientPortNum() == 0) continue; // port # was not set if (!rtspClient->setupMediaSubsession(*subsession)) { *env << "Failed to setup /"" << subsession->mediumName() << "/" << subsession->codecName() << "/" subsession: " << env->getResultMsg() << "/n"; } else { *env << "Setup /"" << subsession->mediumName() << "/" << subsession->codecName() << "/" subsession (client ports " << subsession->clientPortNum() << "-" << subsession->clientPortNum()+1 << ")/n"; } 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 second subsession->rtpSource()->setPacketReorderingThresholdTime(thresh); } } iter.reset(); while ((subsession = iter.next()) != NULL) { if (subsession->readSource() == NULL) continue; // was not initiated char outFileName[1000]; static unsigned streamCounter = 0; snprintf(outFileName, sizeof outFileName, "%s-%s-%d", subsession->mediumName(), subsession->codecName(), ++streamCounter); 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); } 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: unsigned int num=0; SPropRecord * sps=parseSPropParameterSets(subsession->fmtp_spropparametersets(),num); fileSink = H264VideoFileSink::createNew(*env, outFileName,100000); struct timeval tv={0,0}; unsigned char start_code[4] = {0x00, 0x00, 0x00, 0x01}; fileSink->addData(start_code, 4, tv); fileSink->addData(sps[0].sPropBytes,sps[0].sPropLength,tv); fileSink->addData(start_code, 4, tv); fileSink->addData(sps[1].sPropBytes,sps[1].sPropLength,tv); delete[] sps; } else { // Normal case: fileSink = FileSink::createNew(*env, outFileName); } subsession->sink = fileSink; subsession->sink->startPlaying(*(subsession->readSource()),NULL,NULL); } rtspClient->playMediaSession(*session, 0.0f, 0.0f, (float)1.0); env->taskScheduler().doEventLoop(); // does not return return 0; // only to prevent compiler warning }