[gstreamer-bugs] [Bug 352456] Audio plays 1 sec then stops

[GStreamer (bugzilla.gnome.org)]

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http://bugzilla.gnome.org/show_bug.cgi?id=352456
 GStreamer | gst-plugins-base | Ver: 0.10.7


Wim Taymans changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
                 CC|                            |wim at fluendo.com
             Status|UNCONFIRMED                 |RESOLVED
         Resolution|                            |NOTABUG
   Target Milestone|HEAD                        |0.10.7




------- Comment #1 from Wim Taymans  2006-08-23 11:34 UTC -------
Does not seem like a bug, udpsrc is a live source, probably producing data at a
slower rate then needed by alsasink and thus alsasink starvates as buffers are
too late.

The solution is to 

1) set alsasink sync=false so that it tries to follow the rate.
2) use an RTP session manager that exposes a clock to the pipeline based on the
rate of the UDP packets received. alsasink will then slave itself to this clock
and adjust the rate to this clock. This will however not work yet since latency
calculations are not yet implemented, see bug #340842.

The baseaudiosink base class allocates one extra segment so that it has extra
headroom when preparing samples for the audio card.

Please reopen if you suspect something else is wrong.


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