RTSP客户端接收存储数据(live555库中的testRTSPClient实例)
一、testRTSPClient简介html
testRTSPClient是个简单的客户端实例，这个实例对rtsp数据交互做了详细的描述，其中涉及到rtsp会话的两个概念Source和Sink.linux
Source是生产数据，Sink是消费数据. windows
testRTSPClient很是简洁，除了接收服务端发送过来的数据，什么都没干，因此咱们很方便在这个基础上改造，作咱们本身的项目. 缓存
二、testRTSPClient编译，运行session
在linux下编译运行更方便，鉴于个人电脑太渣，虚拟机跑起来费劲，就转到windows下来折腾.app
在windows下只须要加载这一个文件就能够编译，咱们以mediaServer为服务端，以testRTSPClient为客户端。框架
固然也能够用支持rtsp协议的摄像机或其余实体设备做为服务端。ide
先启动mediaServer，而后在testRTSPClient项目的命令菜单里填入mediaServer 提示的IP, 再启动testRTSPClient便可。ui
三、testRTSPClient核心代码解读this
1）看代码以前能够大体浏览一下整体的框架，这位博主画了个流程图http://blog.csdn.net/smilestone_322/article/details/17297817
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http://blog.csdn.net/occupy8/article/details/36426821
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 1 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 //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; }
testRTSPClient接收的fReceiveBuffer缓存没有起始码，start_code[4] = {0x00, 0x00, 0x00, 0x01};