[gstreamer-bugs] [Bug 578506] Pipeline with alsasrc and alsasink cannot change state back to playing

[GStreamer (bugzilla.gnome.org)]

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  GStreamer | gst-plugins-base | Ver: git

Wim Taymans changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
             Status|ASSIGNED                    |RESOLVED
         Resolution|                            |FIXED
   Target Milestone|HEAD                        |0.10.23




------- Comment #2 from Wim Taymans  2009-04-14 11:17 UTC -------
commit dffd1bcc976c432d3827a0a9a34f5ad24edec3d9
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date:   Tue Apr 14 13:16:14 2009 +0200

    baseaudiosrc: adjust the internal timestamp

    Adjust the internal timestamp before comparing it against the adjusted
clock
    time.
    Fixes #578506

commit 0c4c1410f970bca4857acd03527772454dc76e52
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date:   Tue Apr 14 13:12:59 2009 +0200

    baseaudiosink: use new clock time methods

    Use the unadjusted internal clock times to calculate the internal/external
    offset when calibrating the clock.

    When going to NULL, unparent and free the ringbuffer, like we do in the
source
    element.
    See #578506

commit 4231d54823b9811152d9c7e42e293a5e379a3c4b
Author: Wim Taymans <wim.taymans at collabora.co.uk>
Date:   Tue Apr 14 13:08:52 2009 +0200

    audioclock: add methods for the internal offset

    Add two methods for getting the unadjusted time of the clock and one for
    adjusting an internal time. We will need these methods for correctly
handling
    the time after a gst_audio_clock_reset().

    Add a debug category and some debug lines to the audio clock.

    API: gst_audio_clock_get_time()
    API: gst_audio_clock_adjust()
    API: GST_AUDIO_CLOCK_CAST()


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