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
}

转贴：http://blog.csdn.net/zblue78/article/details/5968941