RTSP客戶端接收存儲數據(live555庫中的testRTSPClient實例)
一、testRTSPClient簡介html
testRTSPClient是個簡單的客戶端實例，這個實例對rtsp數據交互做了詳細的描述，其中涉及到rtsp會話的兩個概念Source和Sink.linux
Source是生產數據，Sink是消費數據. windows
testRTSPClient很是簡潔，除了接收服務端發送過來的數據，什麼都沒幹，因此咱們很方便在這個基礎上改造，作咱們本身的項目. 緩存
二、testRTSPClient編譯，運行session
在linux下編譯運行更方便，鑑於個人電腦太渣，虛擬機跑起來費勁，就轉到windows下來折騰.app
在windows下只須要加載這一個文件就能夠編譯，咱們以mediaServer爲服務端，以testRTSPClient爲客戶端。框架
固然也能夠用支持rtsp協議的攝像機或其餘實體設備做爲服務端。ide
先啓動mediaServer，而後在testRTSPClient項目的命令菜單裏填入mediaServer 提示的IP, 再啓動testRTSPClient便可。ui
三、testRTSPClient核心代碼解讀this
1）看代碼以前能夠大體瀏覽一下整體的框架，這位博主畫了個流程圖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）有網友在testRTSPClient基礎上，把接收的數據寫成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緩存沒有起始碼，start_code[4] = {0x00, 0x00, 0x00, 0x01};