<div dir="ltr"><div><div><div>Thanks for your reply. </div>PT 33 is the RTP payload type for MPEG2 TS, which is what I'm expecting. I've attached a log using GST_DEBUG=2,rtp*:6 as requested. </div> </div>I'll put the gst launch crash into a separate message, sorry about that. </div><div class="gmail_extra"> <div class="gmail_quote">On 3 December 2015 at 10:52, Sebastian Dröge <<a href="mailto:sebastian@centricular.com" target="_blank">sebastian@centricular.com</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">On Mi, 2015-12-02 at 13:34 +0000, David Jaggard wrote: > GStreamer 1.6.1 on Windows > > Here is a test pipeline: > gst-launch-1.0.exe rtpbin name=rtpbinout udpsrc caps=application/x- > rtp,media=video,clock-rate=(int)90000 address=239.192.65.121 > port=5000 ! rtpbin ! queue2 ! rtpmp2tdepay ! tsdemux name=demux ! > queue2 ! video/x-h264 ! decodebin ! deinterlace fields=1 method=3 ! > x264enc ! video/x-h264,profile=baseline ! queue2 ! rtph264pay ! > rtpbinout.send_rtp_sink_0 rtpbinout.send_rtp_src_0 ! multiudpsink > clients=<a href="http://227.0.0.82:50090" rel="noreferrer" target="_blank">227.0.0.82:50090</a> ttl-mc=100 demux. ! queue2 ! audio/mpeg ! > decodebin ! audioconvert ! audio/x-raw,channels=1 ! audioresample ! > audio/x-raw,rate=48000 ! voaacenc bitrate=128000 ! > audio/mpeg,mpegversion=4 ! queue2 ! rtpmp4gpay pt=97 ! > rtpbinout.send_rtp_sink_1 rtpbinout.send_rtp_src_1 ! multiudpsink > clients=<a href="http://227.0.0.82:50092" rel="noreferrer" target="_blank">227.0.0.82:50092</a> ttl-mc=100 > > The source mpeg2 ts containing h264 video and aac audio - the > transcoding is necessary for a particular client application. > > Question 1: > Why does it generate a constant stream of these warnings? > rtpsource rtpsource.c:950:calculate_jitter: cannot get clock-rate for > pt 33 That means you get packets from pt=33 somehow but those are not seen as the one you are setting caps for on udpsrc. Just guessing here though, needs at least a debug log with "2,rtp*:6" to be sure. > Question 2: > Why does the pipeline use a GstSystemClock? I would have though it > would clock off the live source. Is this related to question 1? There is nothing else in the pipeline that would provide a clock, so we use the system clock instead. > gst-launch crashes after a few minutes How does it crash? That would be a separate issue. Do you have a backtrace? -- Sebastian Dröge, Centricular Ltd · <a href="http://www.centricular.com" rel="noreferrer" target="_blank">http://www.centricular.com</a> _______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a> <a href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" rel="noreferrer" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote></div> </div>