得到RTP包中的timestamp

NTP------网络时间协议

PTP------精确时间协议

PTS,DTS的关系：

http://www.cnblogs.com/qingquan/archive/2011/07/27/2118967.html

都知道RTSP协议中，真正的数据传输是RTP协议来传输的，每个RTP包都有一个timestamp，（相对时间戳 relative timestamp）这个时间戳是需要经过换算的，我需要把它换算成相应的时间打印到播放器显示的每一帧上。

不过据http://stackoverflow.com/questions/20094998/retrieving-timestamp-in-rtp-rtsp

介绍，AftergettingFrame回调函数中处理的帧时间的，这个时间是PTS（presentationTime timestamp）

以下是代码片断：

 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 > ) envir() << " (with " << numTruncatedBytes << " bytes truncated)";
     char uSecsStr[ + ]; // used to output the 'microseconds' part of the presentation time
     sprintf_s(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);
     envir() << ".\tPresentation time: " << (int)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
     }
 //#ifdef DEBUG_PRINT_NPT
     envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);
 //#endif
     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;
 }

以下是openRTSP Demo的timestamp片断：

  if (notifyTheUser) {
     struct timeval timeNow;
     gettimeofday(&timeNow, NULL);
     char timestampStr[];
     sprintf_s(timestampStr, "%ld%03ld", timeNow.tv_sec, (long)(timeNow.tv_usec/));
     *env << (syncStreams ? "Synchronized d" : "D")
         << "ata packets have begun arriving [" << timestampStr << "]\007\n";
     return;
   }

当然LIVE555自身就在服务端把PTS转换成RTP timestamp了然后进行传输，然后到客户端又自动将RTP timestamp转换成PTS。如果要把PTS转换成世界协调时间（UTS），就需要用gettimeofday()这个函数来转换。

references：

http://comments.gmane.org/gmane.comp.multimedia.live555.devel/12552

http://bbs.csdn.net/topics/390676557?page=1

http://live-devel.live.narkive.com/LfuvIyZj/rtp-timestamp-to-utc-time