[gstreamer-bugs] [Bug 347234] streaming UDP (MPEGTS) shows only one frame of video

[GStreamer (bugzilla.gnome.org)]

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http://bugzilla.gnome.org/show_bug.cgi?id=347234
 GStreamer | gst-plugins-good | Ver: 0.10.6


Wim Taymans changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
             Status|NEW                         |RESOLVED
          Component|don't know                  |gst-plugins-good
         Resolution|                            |FIXED
   Target Milestone|HEAD                        |0.10.4




------- Comment #3 from Wim Taymans  2006-07-12 09:35 UTC -------
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?


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