ChangeLog
=========
2017-06-20 Sebastian Dröge <[email protected]>
* configure.ac:
releasing 1.12.1
2017-06-20 11:08:32 +0300 Sebastian Dröge <[email protected]>
* po/bg.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/fr.po:
* po/hr.po:
* po/hu.po:
* po/nb.po:
* po/pl.po:
* po/ru.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
po: Update translations
2017-06-13 17:40:19 +0300 Vivia Nikolaidou <[email protected]>
* gst/multifile/gstsplitmuxsink.c:
splitmux: Drop allocation queries
They can cause us to deadlock, while we're waiting for a new frame and
upstream is waiting for the allocation query to be answered before
sending a frame
https://bugzilla.gnome.org/show_bug.cgi?id=783753
2017-06-15 10:40:51 +0300 Sebastian Dröge <[email protected]>
* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
rtspsrc: Use a mutex for protecting against concurrent send/receives
We currently send data to the RTSP connection from multiple threads:
whenever a command is to be handled and whenever RTCP is generated.
This
can cause data corruption or worse if both happen at the same time.
As such, protect gst_rtsp_connection_send() and
gst_rtsp_connection_receive()
calls with a mutex. While this means that we hold a mutex during the
IO
operation, this is not actually a problem as the IO operation can be
interrupted (gst_rtsp_connection_flush()) at any time and is blocking
by
itself anyway.
2017-06-15 11:50:44 +0300 Sebastian Dröge <[email protected]>
* gst/isomp4/atoms.c:
qtmux: Un-merge the last two stsc entries after serializing
The last entry will most likely get new samples added to it in
"robust"
muxing mode, changing the samples_per_chunk and thus making it wrong
to
keep the last two entries merged. It will run into an assertion later
when adding a new sample to the chunk.
Thanks to [email protected] for the analysis of the bug and
proposal for a solution.
2017-06-14 00:09:25 +0300 Sebastian Dröge <[email protected]>
* gst/wavparse/gstwavparse.c:
wavparse: Actually clip to upstream size instead of size of the data
chunk
There might be other chunks after the data chunk, so clipping the
chunk
size with the data size can lead to a negative number and all
following
calculations go wrong and cause crashes or worse.
This was introduced in 3ac119bbe2c360e28c087cf3852ea769d611b120.
https://bugzilla.gnome.org/show_bug.cgi?id=783760
2017-05-30 22:23:10 +0200 Juan Navarro <[email protected]>
* gst/rtpmanager/rtpsession.c:
rtpsession: print value of unknown RTCP Payload Type
This adds printing the actual value of any unknown RTCP PT
to the already existing WARNING log message.
https://bugzilla.gnome.org/show_bug.cgi?id=783248
2017-06-02 11:30:15 +0100 Tim-Philipp Müller <[email protected]>
* gst/rtp/gstrtph265depay.c:
rtph265depay: fix caps leak
2017-05-24 11:33:05 +0530 vijay <[email protected]>
* gst/audioparsers/gstaacparse.c:
aacparse : Fix, Caps were not set while reusing aacparse
While reusing aacparse caps were not set.This fix enables aacparse to
reuse in same pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=783027
2017-05-16 12:56:15 +0300 Vivia Nikolaidou <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: Do not check timecode data for mp4 container
Timecode trak is only supported for mov right now, not for mp4. That
code would otherwise create an invalid trak if the muxed video
contained
timecode metadata.
https://bugzilla.gnome.org/show_bug.cgi?id=782684
2017-05-10 15:58:41 +0200 Sebastian Dröge <[email protected]>
* gst/isomp4/gstqtmux.c:
qtmux: Lateness is in QT timescale, diff in GstClockTime
Print the right one in debug output to get meaningful numbers.
2017-05-09 11:41:25 +0200 Sebastian Dröge <[email protected]>
* ext/vpx/gstvpxdec.c:
vpxdec: Set fb->priv to NULL after freeing just in case
https://bugzilla.gnome.org/show_bug.cgi?id=782359
2017-05-08 15:22:00 +0000 Dustin Spicuzza <[email protected]>
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
directsoundsink: Use GstClock API instead of Sleep() for waiting
It's more accurate and allows cancellation.
https://bugzilla.gnome.org/show_bug.cgi?id=773681
2017-05-08 15:05:45 +0000 Tim-Philipp Müller <[email protected]>
* ext/vpx/gstvp9dec.c:
vpx: fix build against older libvpx versions
Such as 1.3.0 as on raspbian.
2017-05-03 23:23:10 +0530 Nirbheek Chauhan <[email protected]>
* sys/directsound/gstdirectsoundsink.c:
directsoundsink: Fix corner case causing large CPU usage
We were unnecessarily looping/goto-ing repeatedly when we had exactly
the amount of data as the free space, and also when the free space was
too small. This, as it turns out, is a very common scenario with
Directsound on Windows.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=773681
We have to do polling here because the event notification API that
Directsound exposes cannot be used with live playback since all events
must be registered in advance with the capture buffer, you cannot
add/remove them once playback has begun. Directsoundsrc had the same
problem.
See also: https://bugzilla.gnome.org/show_bug.cgi?id=781249
Download
========
https://download.gnome.org/sources/gst-plugins-good/1.12/gst-plugins-good-1.12.1.tar.xz
(3.33M)
sha256sum: 121e8e46a7f0e622f09ec9be012607b89d737dd72d48b0f2f0680821ae2cf54b
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