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：
1 void DummySink::afterGettingFrame(void* clientData, unsigned frameSize, unsigned numTruncatedBytes, 2 struct timeval presentationTime, unsigned durationInMicroseconds) { 3 DummySink* sink = (DummySink*)clientData; 4 sink->afterGettingFrame(frameSize, numTruncatedBytes, presentationTime, durationInMicroseconds); 5 } 6 7 // If you don't want to see debugging output for each received frame, then comment out the following line: 8 #define DEBUG_PRINT_EACH_RECEIVED_FRAME 1 9 10 void DummySink::afterGettingFrame(unsigned frameSize, unsigned numTruncatedBytes,11 struct timeval presentationTime, unsigned /*durationInMicroseconds*/) {12 // We've just received a frame of data. (Optionally) print out information about it:13 #ifdef DEBUG_PRINT_EACH_RECEIVED_FRAME14 if (fStreamId != NULL) envir() << "Stream \"" << fStreamId << "\"; ";15 envir() << fSubsession.mediumName() << "/" << fSubsession.codecName() << ":\tReceived " << frameSize << " bytes";16 if (numTruncatedBytes > 0) envir() << " (with " << numTruncatedBytes << " bytes truncated)";17 char uSecsStr[6 + 1]; // used to output the 'microseconds' part of the presentation time18 sprintf_s(uSecsStr, "%06u", (unsigned)presentationTime.tv_usec);19 envir() << ".\tPresentation time: " << (int)presentationTime.tv_sec << "." << uSecsStr;20 if (fSubsession.rtpSource() != NULL && !fSubsession.rtpSource()->hasBeenSynchronizedUsingRTCP()) {21 envir() << "!"; // mark the debugging output to indicate that this presentation time is not RTCP-synchronized22 }23 //#ifdef DEBUG_PRINT_NPT24 envir() << "\tNPT: " << fSubsession.getNormalPlayTime(presentationTime);25 //#endif26 envir() << "\n";27 #endif28 29 // Then continue, to request the next frame of data:30 continuePlaying();31 }32 33 Boolean DummySink::continuePlaying() {34 if (fSource == NULL) return False; // sanity check (should not happen)35 36 // Request the next frame of data from our input source. "afterGettingFrame()" will get called later, when it arrives:37 fSource->getNextFrame(fReceiveBuffer, DUMMY_SINK_RECEIVE_BUFFER_SIZE,38 afterGettingFrame, this,39 onSourceClosure, this);40 return True;41 }
以下是openRTSP Demo的timestamp片断：
1 if (notifyTheUser) {2 struct timeval timeNow;3 gettimeofday(&timeNow, NULL);4 char timestampStr[100];5 sprintf_s(timestampStr, "%ld%03ld", timeNow.tv_sec, (long)(timeNow.tv_usec/1000));6 *env << (syncStreams ? "Synchronized d" : "D")7 << "ata packets have begun arriving [" << timestampStr << "]\007\n";8 return;9 }
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