[gstreamer-bugs] [Bug 419351] [avi/a52dec] AV synchronization problems

[GStreamer (bugzilla.gnome.org)]

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  GStreamer | gst-plugins-base | Ver: 0.10.x

Wim Taymans changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
             Status|NEW                         |RESOLVED
          Component|gst-plugins-good            |gst-plugins-base
         Resolution|                            |FIXED
   Target Milestone|HEAD                        |0.10.20




------- Comment #13 from Wim Taymans  2008-05-09 16:37 UTC -------
very subtle problem caused by rounding errors in the conversion from number of
samples to nanoseconds. The rounding errors accumulated and caused a slight
drift.
Should be fixed now with this patch:

        * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
        Choose to allocate one less segment but require one additional segment
        as latency.

        * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
        No need to increment the number of segments in the source.

        * gst-libs/gst/audio/gstbaseaudiosink.c:
        (gst_base_audio_sink_get_time), (clock_convert_external),
        (gst_base_audio_sink_resample_slaving),
        (gst_base_audio_sink_skew_slaving),
        (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
        (gst_base_audio_sink_async_play):
        Remove adding latency when returning the internal time while
subtracting
        it again when we use the value a little later.
        When calculating the end timestamp, we are making a rounding error
        with the current algorithm. Ensure that we don't accumulate these
        rounding errors when aligning samples by not resampling at all if we
        don't need to. Fixes #
        Make the initial calibration of the clock slaving a little more
        predictable and accurate. Also handle the case where we don't do
        clock slaving.


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