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;

    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
}

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参照openRTSP写的一个RTSP client 加了一些注解

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