<div dir="ltr"><div><div>OK, you say that the stream is RTP with MP2T inside. So you need something like: </div>udpsrc <...> ! 'application/x-rtp,media=video,payload=33,clock-rate=90000,encoding-name=H264'\ ! rtpmp2tdepay ! mpegtsdemux ! more stuff. </div>You probably want a 'T' (tee) to branch off the different programs for their respective decoders. This isn't complete but should get you started. </div><div class="gmail_extra"> <div class="gmail_quote">On Wed, May 7, 2014 at 7:43 AM, antimatter <<a href="mailto:antimatter@quantentunnel.de" target="_blank">antimatter@quantentunnel.de</a>> wrote: <blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello, I am a beginner with GStreamer and haven't done any fancy stuff so far. My goal is to play a RTP multicast stream from our local network. The video should be decoded with Fluendo codec pack (GPU accelerated), because the target platforms have potent GPUs but only Atom CPUs. The general problem is failing sound right after start (manual pipeline) or asynchronous/distorted sound and lots of artifacts (playbin). The stream originates from a DVB-S2 network tuner, which is sending a RTP stream with a MPEG transport stream. Inside there's a H.264 720p video stream, a 2-channel audio stream as AC3 and MP2. The teletext stream is not needed. VLC can play this stream perfectly fine. I provide the stream address "rtp://<a href="http://239.5.2.1:6670" target="_blank">239.5.2.1:6670</a>". Before I start coding, I'd like to build a prototype pipeline with gst-launch-0.10. So I used playbin and provided the URI with "udp" instead of "rtp" and created a diagram from the DOT file, to see what's going on under the hood. The result is having those problems as described at the beginning. Playbin creates an udpsrc, a typefinder, a TSDemux and a multiqueue at the beginning. I recreated this part on my own and added the Fluendo decoder for video and an audio decoder: gst-launch-0.10 -v -m \ udpsrc uri="udp://<a href="http://239.5.2.1:6670" target="_blank">239.5.2.1:6670</a>" ! typefind ! flutsdemux name=demux \ multiqueue name=mq \ demux.video_13ed ! mq.sink0 \ demux.audio_13f2 ! mq.sink1 \ mq.src0 ! fluvadec ! fluvasink \ mq.src1 ! ac3parse ! ffdec_ac3 ! audioconvert ! autoaudiosink This will display the video with artifacts (especially at moving image parts), audio is gone after 1 sec or it is extremely scrambled. Some other forum posts indicate the use of rtpbin, but I could not quite get my head around its usage. And I've got the feeling, this is the way to go. Can I use rtpbin with all it's glory with gst-launch? How do I connect the udpsrc to rtpbin and rtpbin to my decoders? The SSRC and payload type parameters on its dynamic source pad got me lost. And there's something with caps and "application/x-rtp" as mimetype I have seen given to udprsc. Do I need this twice (audio + video)? I was hoping someone could give me the famous push in the right direction. Best regards, Andreas -- View this message in context: <a href="http://gstreamer-devel.966125.n4.nabble.com/Playing-RTP-stream-has-artifacts-and-sound-problems-tp4666829.html" target="_blank">http://gstreamer-devel.966125.n4.nabble.com/Playing-RTP-stream-has-artifacts-and-sound-problems-tp4666829.html</a> Sent from the GStreamer-devel mailing list archive at Nabble.com. _______________________________________________ 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" target="_blank">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a> </blockquote></div> </div>