<div dir="ltr">Hi Nicolas, setting offset on appsrc solved my issue. I really appreciate your input. </div><div class="gmail_extra"> <div class="gmail_quote">On Sat, Jun 18, 2016 at 11:27 AM, Nicolas Dufresne <<a href="mailto:nicolas.dufresne@gmail.com" target="_blank">nicolas.dufresne@gmail.com</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Le jeudi 16 juin 2016 à 17:05 -0400, Jake Zhang a écrit : > Greetings, > > I have been working on this for a while but still have no luck to get > this working. My pipeline can be simplified as below: > > pipeline1: > v4l2src -> tee -> x264enc -> appsink1 > -> appsink2 > > RTSP server pipeline2: > appsrc -> omxh264enc -> rtph264pay > > Pipeline 1 have other processing unit and I will need pull buffers > from appsink2 and push those buffers into appsrc of RTSP server. > My RTSP server part of implementation is very similar to test- > appsrc.c of gst-rtsp-server 1.6.4. > > The buffer I got from appsink already has their own PTS and to get > RTSP pipeline2 running, I have to re-stamp the buffer PTS the exactly > same way as test-appsrc.c (set ctx->timestamp=0 and increment based > of buffer duration) otherwise, the RTSP server will not get running. You don't, you can use push_sample(), so the segment is passed. And then, on the first buffer, you can compute the running time of this buffer (gst_segment_to_running_time()), and set the appsrc pad offset to -running_time_first_pts. Careful, that if you add audio, you should use the same offset for all stream, otherwise you will break a/v sync. regards, Nicolas _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> <a href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote></div> </div>