[gstreamer-bugs] [Bug 347234] streaming UDP (MPEGTS) shows only one frame of video
GStreamer (bugzilla.gnome.org)
bugzilla-daemon at bugzilla.gnome.org
Wed Jul 12 02:35:18 PDT 2006
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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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