gst-plugins-base 1.4.2

From: Sebastian Dröge <install-module master gnome org>
To: FTP Releases <ftp-release-list gnome org>
Subject: gst-plugins-base 1.4.2
Date: Fri, 19 Sep 2014 12:21:53 +0000 (UTC)
ChangeLog
=========

2014-09-19  Sebastian Dröge <slomo coaxion net>

        * configure.ac:
          releasing 1.4.2

2014-09-19 09:58:48 +0300  Sebastian Dröge <sebastian centricular com>

        * po/da.po:
        * po/sr.po:
          po: Update translations

2014-09-18 12:29:37 +0400  Andrei Sarakeev <sarakusha gmail com>

        * gst/playback/gstplaybin2.c:
          playbin: Don't leak input-selector sinkpads
          https://bugzilla.gnome.org/show_bug.cgi?id=736861

2014-09-17 14:18:49 +0200  Ognyan Tonchev <ognyan axis com>

        * gst-libs/gst/audio/gstaudioencoder.c:
          audioencoder: do not leak events when flushing them
          https://bugzilla.gnome.org/show_bug.cgi?id=736796

2014-09-17 14:34:25 +0200  Ognyan Tonchev <ognyan axis com>

        * gst/encoding/gststreamsplitter.c:
          streamsplitter: do not leak events when flushing them
          https://bugzilla.gnome.org/show_bug.cgi?id=736796

2014-09-17 14:11:21 +0200  Ognyan Tonchev <ognyan axis com>

        * gst-libs/gst/video/gstvideodecoder.c:
          videodecoder: do not leak events when flushing them
          https://bugzilla.gnome.org/show_bug.cgi?id=736796

2014-09-18 12:39:48 +0300  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/audio/gstaudiodecoder.c:
          audiodecoder: Simplify code a bit

2014-09-17 14:08:17 +0200  Ognyan Tonchev <ognyan axis com>

        * gst-libs/gst/video/gstvideoencoder.c:
          videoencoder: do not leak events when flushing them
          https://bugzilla.gnome.org/show_bug.cgi?id=736796

2014-09-17 12:17:53 +0200  Ognyan Tonchev <ognyan axis com>

        * gst-libs/gst/audio/gstaudiodecoder.c:
          audiodecoder: Don't leak events
          https://bugzilla.gnome.org/show_bug.cgi?id=736788

2014-09-17 12:17:27 +0200  Ognyan Tonchev <ognyan axis com>

        * tests/check/libs/audiodecoder.c:
          audiodecoder: extend flush_events test to check for event leaks
          https://bugzilla.gnome.org/show_bug.cgi?id=736788

2014-09-05 13:49:46 -0300  Thiago Santos <thiagoss osg samsung com>

        * ext/pango/gstbasetextoverlay.c:
          basetextoverlay: Do not fail the negotiation if query fails
          The allocation query failure doesn't mean that the negotiation
          has failed as the element can allocate buffers itself.
          Instead, only fail if the pads are flushing and the allocation
          query failed.
          https://bugzilla.gnome.org/show_bug.cgi?id=735844

2013-01-31 13:49:00 +0100  Arnaud Vrac <avrac freebox fr>

        * ext/pango/gstbasetextoverlay.c:
          basetextoverlay: get framerate from previously parsed video info

2013-01-31 13:47:35 +0100  Arnaud Vrac <avrac freebox fr>

        * ext/pango/gstbasetextoverlay.c:
          basetextoverlay: do not ask for a bufferpool when checking for composition meta

2014-09-04 15:06:31 +0200  Arnaud Vrac <avrac freebox fr>

        * ext/pango/gstbasetextoverlay.c:
          basetextoverlay: schedule reconfigure on source pad when negotiation fails
          The source pad might be flushing while negotiating, resulting in
          set_caps or the ALLOCATION query failing. In this case set the
          reconfigure flag on the source pad so that negotiation is retried on the
          next buffer.

2014-09-16 13:32:52 +0200  Ognyan Tonchev <ognyan axis com>

        * gst-libs/gst/audio/gstaudiocdsrc.c:
          audiocdsrc: do not leak uid after parsing TOC select event
          https://bugzilla.gnome.org/show_bug.cgi?id=736739

2014-09-17 10:51:59 +0530  Ravi Kiran K N <ravi kiran samsung com>

        * gst/typefind/gsttypefindfunctions.c:
          typefind: correct the condition for irap flag
          https://bugzilla.gnome.org/show_bug.cgi?id=736779

2014-09-16 21:42:46 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstplaysink.c:
          playsink: Add audio/videoconvert in front of the audio/video-filters
          audioresample and videoscale is something the application will have to do if
          required, but we can at least help here by adding the
          audioconvert/videoconvert elements.
          https://bugzilla.gnome.org/show_bug.cgi?id=735748

2014-09-15 16:23:57 +0200  Ognyan Tonchev <ognyan axis com>

        * gst-libs/gst/video/gstvideodecoder.c:
          videodecoder: do not leak pool and allocator in error case
          https://bugzilla.gnome.org/show_bug.cgi?id=736679

2014-09-05 09:54:10 -0700  Garg <aksg86 gmail com>

        * gst-libs/gst/audio/gstaudiobasesink.c:
          audiobasesink: Fix deadlock caused by holding object lock while calling clock functions
          Issue:
          During a PAUSED->PLAYING transition when we are rendering an audio buffer in AudioBaseSink
          we make adjustments to the sink's provided clock i.e. fix clock calibration using the external
          pipeline clock, within "gst_audio_base_sink_sync_latency function inside gstaudiobasesink.c".
          For the calibration adjustment we need to get the sink clock time using "gst_audio_clock_get_time".
          But before calling "gst_audio_clock_get_time" we acquire the Object Lock on the Sink. If sink is
          a pulsesink, "gst_audio_clock_get_time" internally calls "gst_pulsesink_get_time" which needs to
          acquire Pulse Audio Main Loop Lock before querying Pulse Audio for its stream time using
          "pa_stream_get_time". Please see "gst_pulsesink_get_time in pulsesink.c".
          So the situation here is we have acquired the Object lock on Sink and need PA Main Loop Lock.
          Now Pulse Audio Main Thread itself might be in the process of posting a stream status
          message after Paused to Playing transition which in turn acquires the PA Main loop lock and
          needs the Object Lock on Pulse Sink. This causes a deadlock with the earlier render thread.
          Fix:
          Do not acquire the object Lock on Sink before querying the time on PulseSink clock. This is
          similar to the way we have used get_time at other places in the code. Acquire it after the
          get_time call. This way PA Main loop will be able to post its stream status message by
          acquiring the Sink Object lock and will eventually release its Main Loop lock needed for
          gst_pulsesink_get_time to continue.
          https://bugzilla.gnome.org/show_bug.cgi?id=736071

2014-09-12 14:27:44 +0300  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/video/gstvideofilter.c:
          videofilter: Unref buffers before calling the transform_frame functions
          GstVideoFrame has another reference, so the buffer looks unwriteable,
          meaning that we can't attach any metas or anything to it
          https://bugzilla.gnome.org/show_bug.cgi?id=736118

2014-09-11 16:58:35 -0300  Thiago Santos <thiagoss osg samsung com>

        * gst/playback/gstdecodebin2.c:
          decodebin: protect buffering message handling
          Use the object lock to avoid concurrent processing which leads
          to small disasters (assertions or crashes)

2014-03-28 13:02:54 +0100  George Kiagiadakis <george kiagiadakis collabora com>

        * gst/playback/gstplaybin2.c:
          playbin: filter out buffering messages when switching uri
          When switching URI from about-to-finish, playbin starts decoding the new
          URI and the queue2 inside uridecodebin starts emitting buffering messages
          immediately. However, the queue(s) inside playsink still have buffers to
          play and the pipeline doesn't need to pause for buffering, so we should
          not send those buffering messages up to the application, otherwise there
          is an audible glitch caused by pausing the pipeline for a very short time.
          https://bugzilla.gnome.org/show_bug.cgi?id=727255

2014-07-08 12:37:41 -0400  Kipp Cannon <kipp cannon ligo org>

        * gst/audioresample/resample.c:
          audioresample: don't skip input samples
          when downsampling, the output buffer can be filled before all the input
          samples are consumed.  this is correct:  when downsampling, several input
          samples are needed for each output sample, so when only a small number of
          input samples are available the number of output samples produced can be 0.
          the resampler, however, was discarding those extra input samples instead of
          clocking them into its filter history for the next iteration.  this patch
          fixes this by removing the check that the output buffer is full.  the code
          now always loops until all input samples are consumed, and relies on the
          calling code to have provided a suitably sized location for the output.
          note that there are already other checks in place in the calling code to
          ensure that this is the case.
          https://bugzilla.gnome.org/show_bug.cgi?id=732908

2014-08-27 13:45:57 +0200  Göran Jönsson <goranjn axis com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtspconnection: Protect readsrc, writesrc and controllsrc with a mutex
          Fixes a crash when controlsrc, readsrc or writesrc are modified from
          gst_rtsp_source_dispatch_read/write and gst_rtsp_watch_reset at the
          same time.
          https://bugzilla.gnome.org/show_bug.cgi?id=735569

2014-09-03 15:23:26 +0530  Vineeth T M <vineeth tm samsung com>

        * gst/videorate/gstvideorate.c:
          videorate: GstStructure refcount critical message
          s3 is not being initialized when run in a loop
          and the same was being freed, which resulted in the crash
          https://bugzilla.gnome.org/show_bug.cgi?id=735952

2014-09-01 15:23:27 -0300  Thiago Santos <thiagoss osg samsung com>

        * tests/check/elements/textoverlay.c:
          tests: textoverlay: add test to reproduce fakesink scenario
          Adds a new test to textoverlay to make sure it can properly handle
          elements that have ANY caps but fail to add the overlay meta in
          the allocation query.
          This test verifies that textoverlay won't use the caps features even
          knowing that the overlay meta is accepted when querying the downstream
          caps because it also needs downstream to confirm by putting the meta
          in the allocation query.
          https://bugzilla.gnome.org/show_bug.cgi?id=735800

2014-09-01 12:38:02 -0300  Thiago Santos <thiagoss osg samsung com>

        * ext/pango/gstbasetextoverlay.c:
          basetextoverlay: properly fallback to non-overlay caps
          When downstream claims to accept the overlay meta but fails to
          provide it in the allocation query, properly fallback to setting
          a new caps without the overlay meta as that is not going to be used.
          Only do this if the original caps doesn't have the overlay already,
          otherwise there isn't much that can be done.
          https://bugzilla.gnome.org/show_bug.cgi?id=735800

2014-09-01 12:28:24 +0300  Sebastian Dröge <sebastian centricular com>

        * ext/pango/gstbasetextoverlay.c:
          textoverlay: Don't hold any mutexes while calling negotiate
          It's not done in any other code calling negotiate and will cause deadlocks
          as it is sending events and queries in the pipeline.
          Specifically this pipeline was deadlocking:
          gst-launch-1.0 videotestsrc ! textoverlay ! textoverlay ! fakesink



Download
========
https://download.gnome.org/sources/gst-plugins-base/1.4/gst-plugins-base-1.4.2.tar.xz (2.50M)
  sha256sum: c0a8c44607d8a5669d2f0c118a72026f883a58ce1f3c720924b77f275b7b8835
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