RTSP客戶端接收存儲數據(live555庫中的testRTSPClient實例)

1、testRTSPClient簡介

testRTSPClient是個簡單的客戶端實例，這個實例對rtsp數據交互作了詳細的描述，其中涉及到rtsp會話的兩個概念Source和Sink.

Source是生產數據，Sink是消費數據.  

testRTSPClient非常簡潔，除了接收服務端發送過來的數據，什么都沒干，所以我們很方便在這個基礎上改造，做我們自己的項目. 

 

2、testRTSPClient編譯，運行

在linux下編譯運行更方便，鑒於我的電腦太渣，虛擬機跑起來費勁，就轉到windows下來折騰.

在windows下只需要加載這一個文件就可以編譯，我們以mediaServer為服務端，以testRTSPClient為客戶端。

當然也可以用支持rtsp協議的攝像機或其他實體設備作為服務端。

 

先啟動mediaServer，然后在testRTSPClient項目的命令菜單里填入mediaServer 提示的IP, 再啟動testRTSPClient即可。

 

3、testRTSPClient核心代碼解讀

1）看代碼之前可以大致瀏覽一下總體的框架，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;
}

 

3）testRTSPClient這個實例還支持多路錄放，網上搜到有人已經實現了，搬過來.

     http://blog.chinaunix.net/uid-15063109-id-4482932.html