1、testRTSPClient简介

testRTSPClient是个简单的客户端实例，这个实例对rtsp数据交互作了详细的描述，其中涉及到rtsp会话的两个概念Source和Sink.

Source是生产数据，Sink是消费数据.  

testRTSPClient非常简洁，除了接收服务端发送过来的数据，什么都没干，所以我们很方便在这个基础上改造，做我们自己的项目. 

 

2、testRTSPClient编译，运行

在linux下编译运行更方便，鉴于我的电脑太渣，虚拟机跑起来费劲，就转到windows下来折腾.

在windows下只需要加载这一个文件就可以编译，我们以mediaServer为服务端，以testRTSPClient为客户端。

当然也可以用支持rtsp协议的摄像机或其他实体设备作为服务端。

 

先启动mediaServer，然后在testRTSPClient项目的命令菜单里填入mediaServer 提示的IP, 再启动testRTSPClient即可。

 

3、testRTSPClient核心代码解读

1）看代码之前可以大致浏览一下总体的框架，这位博主画了个流程图

void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,                  struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {  // We've just received a frame of data.  (Optionally) print out information about it:#ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME  if (fStreamId != NULL) envir() << "Stream \"" << fStreamId << "\"; ";  envir() << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";  if (numTruncatedBytes > 0) envir() << " (with " << numTruncatedBytes << " bytes truncated)";  char uSecsStr[6+1]; // used to output the 'microseconds' part of the presentation time  sprintf(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);  envir() << ".\tPresentation time: " << (unsigned)presentationTime.tv_sec << "." << uSecsStr;  if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {    envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized  }  envir() << "\n";#endif    // Then continue, to request the next frame of data:  continuePlaying();}Boolean DummySink::continuePlaying() {  if (fSource == NULL) return False; // sanity check (should not happen)  // Request the next frame of data from our input source.  "afterGettingFrame()" will get called later, when it arrives:  fSource->getNextFrame(fReceiveBuffer, DUMMY_SINK_RECEIVE_BUFFER_SIZE,                        afterGettingFrame, this,                        onSourceClosure, this);  return True;}

 

2）有网友在testRTSPClient基础上，把接收的数据写成h264文件了 

void DummySink::afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes,                  struct timeval presentationTime, unsigned durationInMicroseconds) {  DummySink* sink = (DummySink*)clientData;  sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime, durationInMicroseconds);}// If you don't want to see debugging output for each received frame, then comment out the following line:#define DEBUG_PRINT_EACH_RECEIVED_FRAME 1void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes, struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {  // We've just received a frame of data.  (Optionally) print out information about it:#ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME  if (fStreamId != NULL) envir() << "Stream \"" << fStreamId << "\"; ";  envir() << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";  if (numTruncatedBytes > 0) envir() << " (with " << numTruncatedBytes << " bytes truncated)";  char uSecsStr[6+1]; // used to output the 'microseconds' part of the presentation time  sprintf(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);  envir() << ".\tPresentation time: " << (unsigned)presentationTime.tv_sec << "." << uSecsStr;  if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {    envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized  }  envir() << "\n";#endif    //todo one frame  //save to file  if(!strcmp(fSubsession.mediumName(), "video"))  {     if(firstFrame)     {         unsigned int num;         SPropRecord *sps = parseSPropParameterSets(fSubsession.fmtp_spropparametersets(), num);         // For H.264 video stream, we use a special sink that insert start_codes:         struct timeval tv= {0,0};         unsigned char start_code[4] = {0x00, 0x00, 0x00, 0x01};         FILE *fp = fopen("test.264", "a+b");         if(fp)         {             fwrite(start_code, 4, 1, fp);             fwrite(sps[0].sPropBytes, sps[0].sPropLength, 1, fp);             fwrite(start_code, 4, 1, fp);             fwrite(sps[1].sPropBytes, sps[1].sPropLength, 1, fp);             fclose(fp);             fp = NULL;         }         delete [] sps;         firstFrame = False;     }     char *pbuf = (char *)fReceiveBuffer;     char head[4] = {0x00, 0x00, 0x00, 0x01};     FILE *fp = fopen("test.264", "a+b");     if(fp)     {         fwrite(head, 4, 1, fp);         fwrite(fReceiveBuffer, frameSize, 1, fp);         fclose(fp);         fp = NULL;     }  }  // Then continue, to request the next frame of data:  continuePlaying();}Boolean DummySink::continuePlaying() {  if (fSource == NULL) return False; // sanity check (should not happen)  // Request the next frame of data from our input source.  "afterGettingFrame()" will get called later, when it arrives:  fSource->getNextFrame(fReceiveBuffer, DUMMY_SINK_RECEIVE_BUFFER_SIZE,                        afterGettingFrame, this,                        onSourceClosure, this);  return True;}

 

3）testRTSPClient这个实例还支持多路录放，网上搜到有人已经实现了，搬过来.