参照openRTSP写的一个RTSP client

 

1. #include "liveMedia.hh"   
2. #include "BasicUsageEnvironment.hh"   
3. #include "GroupsockHelper.hh"   
4. UsageEnvironment* env;  
5. portNumBits tunnelOverHTTPPortNum = 0;  
6. const char * url="rtsp://127.0.0.1:1935/vod/Extremists.m4v";  
7. #if defined(__WIN32__) || defined(_WIN32)   
8. #define snprintf _snprintf   
9. #endif   
10. int main(int argc,const char ** argv)  
11. {  
12.     //创建BasicTaskScheduler对象   
13.     TaskScheduler* scheduler = BasicTaskScheduler::createNew();  
14.     //创建BisicUsageEnvironment对象   
15.     env = BasicUsageEnvironment::createNew(*scheduler);  
16.     //创建RTSPClient对象   
17.     RTSPClient * rtspClient= RTSPClient::createNew(*env);  
18.     //由RTSPClient对象向服务器发送OPTION消息并接受回应   
19.     char* optionsResponse=rtspClient->sendOptionsCmd(url);  
20.     delete [] optionsResponse;  
21.     //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）   
22.     char* sdpDescription =rtspClient->describeURL(url);  
23.     //创建MediaSession对象（根据SDPDescription在MediaSession中创建和初始化MediaSubSession子会话对象）   
24.     MediaSession* session = MediaSession::createNew(*env, sdpDescription);  
25.     delete[] sdpDescription;  
26. /* 
27. while循环中配置所有子会话对象 
28. */  
29.     MediaSubsessionIterator iter(*session);  
30.     MediaSubsession *subsession;  
31.     while ((subsession = iter.next()) != NULL) {  
32.         // Creates a "RTPSource" for this subsession. (Has no effect if it's   
33.         // already been created.)  Returns True iff this succeeds.   
34.         if (!subsession->initiate()) {  
35.             *env << "Unable to create receiver for /"" << subsession->mediumName()  
36.                 << "/" << subsession->codecName()  
37.                 << "/" subsession: " << env->getResultMsg() << "/n";  
38.         } else {  
39.             *env << "Created receiver for /"" << subsession->mediumName()  
40.                 << "/" << subsession->codecName()  
41.                 << "/" subsession (client ports " << subsession->clientPortNum()  
42.                 << "-" << subsession->clientPortNum()+1 << ")/n";  
43.             if (subsession->rtpSource() != NULL) {  
44.                 // Because we're saving the incoming data, rather than playing   
45.                 // it in real time, allow an especially large time threshold   
46.                 // (1 second) for reordering misordered incoming packets:   
47.                 unsigned const thresh = 1000000; // 1 second   
48.                 subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);  
49.                 // Set the RTP source's OS socket buffer size as appropriate - either if we were explicitly asked (using -B),   
50.                 // or if the desired FileSink buffer size happens to be larger than the current OS socket buffer size.   
51.                 // (The latter case is a heuristic, on the assumption that if the user asked for a large FileSink buffer size,   
52.                 // then the input data rate may be large enough to justify increasing the OS socket buffer size also.)   
53.                 int socketNum = subsession->rtpSource()->RTPgs()->socketNum();  
54.                 unsigned curBufferSize = getReceiveBufferSize(*env, socketNum);  
55.                 unsigned newBufferSize = setReceiveBufferTo(*env, socketNum, 100000);  
56.   
57.             }  
58.         }  
59.     }  
60.     //由RTSPClient对象向服务器发送SETUP消息并接受回应   
61.     iter.reset();  
62.     while ((subsession = iter.next()) != NULL) {  
63.         if (subsession->clientPortNum() == 0) continue; // port # was not set   
64.         if (!rtspClient->setupMediaSubsession(*subsession)) {  
65.             *env << "Failed to setup /"" << subsession->mediumName()  
66.                 << "/" << subsession->codecName()  
67.                 << "/" subsession: " << env->getResultMsg() << "/n";  
68.         } else {  
69.             *env << "Setup /"" << subsession->mediumName()  
70.                 << "/" << subsession->codecName()  
71.                 << "/" subsession (client ports " << subsession->clientPortNum()  
72.                 << "-" << subsession->clientPortNum()+1 << ")/n";  
73.         }  
74.         if (subsession->rtpSource() != NULL) {  
75.             // Because we're saving the incoming data, rather than playing   
76.             // it in real time, allow an especially large time threshold   
77.             // (1 second) for reordering misordered incoming packets:   
78.             unsigned const thresh = 1000000; // 1 second   
79.             subsession->rtpSource()->setPacketReorderingThresholdTime(thresh);  
80.         }  
81.     }  
82.     iter.reset();  
83.     while ((subsession = iter.next()) != NULL) {  
84.         if (subsession->readSource() == NULL) continue; // was not initiated   
85.         char outFileName[1000];  
86.         static unsigned streamCounter = 0;  
87.         snprintf(outFileName, sizeof outFileName, "%s-%s-%d",  
88.             subsession->mediumName(),  
89.             subsession->codecName(), ++streamCounter);  
90.         FileSink* fileSink;  
91.         if (strcmp(subsession->mediumName(), "audio") == 0 &&  
92.             (strcmp(subsession->codecName(), "AMR") == 0 ||  
93.             strcmp(subsession->codecName(), "AMR-WB") == 0)) {  
94.                 // For AMR audio streams, we use a special sink that inserts AMR frame hdrs:   
95.                 fileSink = AMRAudioFileSink::createNew(*env, outFileName);  
96.         } else if (strcmp(subsession->mediumName(), "video") == 0 &&  
97.             (strcmp(subsession->codecName(), "H264") == 0)) {  
98.                 // For H.264 video stream, we use a special sink that insert start_codes:   
99.                 unsigned int num=0;  
100.                 SPropRecord * sps=parseSPropParameterSets(subsession->fmtp_spropparametersets(),num);  
101.                 fileSink = H264VideoFileSink::createNew(*env, outFileName,100000);  
102.                 struct timeval tv={0,0};  
103.                 unsigned char start_code[4] = {0x00, 0x00, 0x00, 0x01};  
104.                 fileSink->addData(start_code, 4, tv);  
105.                 fileSink->addData(sps[0].sPropBytes,sps[0].sPropLength,tv);  
106.                 fileSink->addData(start_code, 4, tv);  
107.                 fileSink->addData(sps[1].sPropBytes,sps[1].sPropLength,tv);  
108.                 delete[] sps;  
109.         } else {  
110.             // Normal case:   
111.             fileSink = FileSink::createNew(*env, outFileName);  
112.         }  
113.         subsession->sink = fileSink;  
114.         subsession->sink->startPlaying(*(subsession->readSource()),NULL,NULL);  
115.     }  
116.     rtspClient->playMediaSession(*session, 0.0f, 0.0f, (float)1.0);  
117.     env->taskScheduler().doEventLoop(); // does not return   
118.     return 0; // only to prevent compiler warning   
119. }