ChangeLog
=========
2014-09-19 Sebastian Dröge <[email protected]>
* configure.ac:
releasing 1.4.2
2014-09-19 09:58:48 +0300 Sebastian Dröge <[email protected]>
* po/da.po:
* po/sr.po:
po: Update translations
2014-09-18 12:29:37 +0400 Andrei Sarakeev <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* gst-libs/gst/audio/gstaudiodecoder.c:
audiodecoder: Simplify code a bit
2014-09-17 14:08:17 +0200 Ognyan Tonchev <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* ext/pango/gstbasetextoverlay.c:
basetextoverlay: get framerate from previously parsed video info
2013-01-31 13:47:35 +0100 Arnaud Vrac <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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 <[email protected]>
* 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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