GNOME Bugzilla – Bug 347234
streaming UDP (MPEGTS) shows only one frame of video
Last modified: 2006-07-18 09:44:14 UTC
When streaming an MPEG1 file with VLC over UDP, trying to play it with gst-launch-0.10 playbin uri=udp://0.0.0.0:1234 only one frame of video is shown. The stream can be played with vlc and mplayer, so the problem is at least partially in gstreamer. I'm using Ubuntu Dapper and have the following gstreamer packages installed: gstreamer0.10-alsa gstreamer0.10-doc gstreamer0.10-esd gstreamer0.10-ffmpeg gstreamer0.10-fluendo-mpegdemux gstreamer0.10-gl gstreamer0.10-gnomevfs gstreamer0.10-pitfdll gstreamer0.10-plugins-bad gstreamer0.10-plugins-bad-multiverse gstreamer0.10-plugins-base gstreamer0.10-plugins-base-apps gstreamer0.10-plugins-good gstreamer0.10-plugins-ugly gstreamer0.10-plugins-ugly-multiverse gstreamer0.10-tools gstreamer0.10-x Attached is the level 5 log. gstreamer freezes after line 280968, so the stuff before that is probably the interesting part.
Looks like the log is too big to attach, so find it at http://foolip.org/udp-mpegts-1frame.complete.log.bz2 (the webserver is misconfigured, so right click to save).
audio decoding fails because the data sent by the server is MPEG TS over RTP. We don't have an RTP depacketizer for MPEG TS yet.
Commited an mpeg2 TS depayloader: * gst/rtp/Makefile.am: * gst/rtp/gstrtp.c: (plugin_init): * gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_base_init), (gst_rtp_mp2t_depay_class_init), (gst_rtp_mp2t_depay_init), (gst_rtp_mp2t_depay_setcaps), (gst_rtp_mp2t_depay_process), (gst_rtp_mp2t_depay_set_property), (gst_rtp_mp2t_depay_get_property), (gst_rtp_mp2t_depay_change_state), (gst_rtp_mp2t_depay_plugin_init): * gst/rtp/gstrtpmp2tdepay.h: Added mpeg2 TS depayloader. Closing #347234. Following line works somewhat: gst-launch-0.10 -v udpsrc uri=udp://localhost:1234 ! application/x-rtp, media=video, clock-rate=90000 ! rtpmp2tdepay ! flutsdemux name=d ! queue ! flumpeg2vdec ! xvimagesink sync=false d. ! queue ! flump3dec ! alsasink sync=false Constructing a good RTP session from a random UDP source is not yet implemented in playbin. Possibly add typefind code for RTP specific payloads?
*** Bug 336752 has been marked as a duplicate of this bug. ***