Hi Daniel,

First, you should ensure your time between audio and video are sync.

Then, just put struct timval into fPresentation time(need sec and usec).


Second, ensure your audio duration time is ok, you could refer ATDS source code.


I hope this may help you. Thanks?.


regards, eric, 11/04


________________________________
從: live-devel <live-devel-boun...@ns.live555.com> 代表 Daniel Yacouboff 
<danie...@essence-grp.com>
寄件日期: 2015年11月4日 下午 06:56
至: live-devel@lists.live555.com
主旨: [Live-devel] Live video and audio streaming using one RTSPServer

Hello there,
I’ve been working on your library for a while now, in order to implement a RTSP 
server, streaming live audio and video.
The audio and video packets are received via TCP from another service, each 
type of packet (audio\video) to another socket.
What I did was creating a new sub-classes of MediaSubsession, one for PCM audio 
and one for H264 video, and for each one created a new framed source, similar 
to ByteStreamFileSource, where in their doReadFromFile() I call recv() to the 
relevant socket instead of fread().
In my main, I just open the TCP sockets, create an RTSP server and add the two 
subsessions to it.

It works fine when I play video alone, or audio alone, but when I combine them 
the video usually stuck after a short while (~30 seconds) and the audio keeps 
playing ok.
In addition, during the time both of them played ok, they are not synched.

What am I doing wrong? I’ve read a lot about it and it seems my presentation 
times are not ok, but how can I calculate them properly?
Please advise, I’ve been working on this forever, and I’m sure it’ll be useful 
for others as well.

Thanks.

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