[Live-devel] Regarding the h264 video stream's rtp packet timestamp

[Tony]

But if I use the vlc to play it, the audio's pts is out of range, and it's dropped.
but it works by ffplay.
I don't know where is wrong.
Could you help me to check it where is wrong. I'm crazy caused by this problem.
Here is my code.
audio is g711, each buffer is 160 bytes.
void AudioFrameSource::doGetNextFrame()
{
    unsigned acquiredFrameSize=0;
    if(m_session!=NULL)
  {
      m_session->GetNextAudioFrame((char*)fTo,fMaxSize,&acquiredFrameSize,&fNumTruncatedBytes);
      if(acquiredFrameSize!=0)
      {
          if(_isFirst)
            {
                m_session->GetTimeScale(&_timeval);
                _isFirst = false;
            }
            else
            {
                _timeval.tv_usec += 20000;
                if(_timeval.tv_usec>=1000000)
                {
                    _timeval.tv_sec ++;
                    _timeval.tv_usec -= 1000000;
                }
            }
            fFrameSize = acquiredFrameSize;
            fPresentationTime = _timeval;
           // fDurationInMicroseconds = 20000;
      }
  }

    nextTask() = envir().taskScheduler().scheduleDelayedTask(20000,(TaskFunc*)FramedSource::afterGetting, this);
}


video's getnextframe
each buffer is  h264 nalu, and the stream's average framerate is 25fps.
void VideoFrameSource::doGetNextFrame()
{
  unsigned int framesize=0;
  if(m_session!=NULL)
  {
      bool lastnalu=false;
      m_session->GetNextVideoFrame(_firstframe,lastnalu,(char*)fTo,fMaxSize,&framesize,&fNumTruncatedBytes);
      fFrameSize = framesize;
      if(framesize!=0)
      {
          if(_firstframe)
          {
               m_session->GetTimeScale(&_timescale);
               _firstframe = false;
          }
          else if(lastnalu)
          {
                _timescale.tv_usec += 40000;
                if(_timescale.tv_usec>=1000000)
                {
                    _timescale.tv_sec ++;
                    _timescale.tv_usec -= 1000000;
                }
          }
        fPresentationTime = _timescale;
       // fDurationInMicroseconds = 40000;
      }
  }

   nextTask() = envir().taskScheduler().scheduleDelayedTask(8000,(TaskFunc*)FramedSource::afterGetting, this);
}



At 2013-10-24 16:35:56,"Ross Finlayson" <finlayson at live555.com> wrote:

I saw the FAQ in live555 website.
It said the live555 sync the audio and video by RTCP's SR packets. 
So I should create RTCP instance for each RTP source explititly?



No, because (assuming that you are controlling the streaming using RTSP) this is done implicitly.


(In the RTSP server, this is done when the stream starts playing; in the RTSP client, it is done in the implementation of the "initiate()" function.)



Ross Finlayson
Live Networks, Inc.
http://www.live555.com/

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