ChangeLog
=========
2017-07-14 Sebastian Dröge <sl...@coaxion.net>
* configure.ac:
releasing 1.12.2
2017-07-14 13:22:45 +0300 Sebastian Dröge <sebast...@centricular.com>
* po/el.po:
po: Update translations
2017-07-13 12:47:02 +0300 Sebastian Dröge <sebast...@centricular.com>
* gst/isomp4/qtdemux.c:
qtdemux: Fix parsing of RLE depth
Regression introduced by 86b427dc70562f891a551ffc9f96cefe1cafcddd
https://bugzilla.gnome.org/show_bug.cgi?id=784812
2017-05-20 17:09:52 +0200 Josep Torra <jto...@oblong.com>
* sys/osxaudio/gstosxcoreaudio.c:
osxaudio: fixes playback of mono streams with no channel-mask field
in caps
Fixes a negotiation error seen when trying to playback of a .MOV file
with
a mono AAC audio stream decoded by avcdec_aac that doesn't set
channel-mask
field but sink was requiring channel-mask=0x3.
2017-07-07 21:15:57 +0900 Yasushi SHOJI <ya...@atmark-techno.com>
* gst/rtp/gstrtpgsmpay.c:
rtpgsmpay: fix accidental garbage data before actual payload
Do not allocate payload size outbuf if appending payload buffer.
The commit 137672ff1824948bda4b1b1967de8c24a0055b67 attached payload
to the output buffer but forgot to remove payload allocation. That
effectively doubled payload size and add zero'ed or random bytes.
Makes the following pipeline work again:
gst-launch-1.0 -v audiotestsrc wave=2 ! gsmenc ! rtpgsmpay !
rtpgsmdepay ! gsmdec ! autoaudiosink
https://bugzilla.gnome.org/show_bug.cgi?id=784616
2017-07-03 11:47:13 -0400 Nicolas Dufresne <nicolas.dufre...@collabora.com>
* gst/rtpmanager/gstrtprtxreceive.c:
rtprtxreceive: Add memory and boundary checks
This element was not checking if mapping the RTP buffer and the
payload
worked, and was not checking if the RTX payload was large enough.
https://bugzilla.gnome.org/show_bug.cgi?id=784484
2017-07-03 20:27:29 +0100 Tim-Philipp Müller <t...@centricular.com>
* gst/imagefreeze/gstimagefreeze.c:
imagefreeze: fix use-after-free on seek event
Get seqnum before unreffing the seek event.
https://bugzilla.gnome.org/show_bug.cgi?id=784486
2017-06-29 18:59:58 +0300 Sebastian Dröge <sebast...@centricular.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Create send/recv mutexes once, not on every connect()
Also fixes a crash caused by freeing an uninitialized mutex in an
error
case.
https://bugzilla.gnome.org//show_bug.cgi?id=784282
2017-06-22 11:38:56 +0300 Sebastian Dröge <sebast...@centricular.com>
* gst/rtsp/gstrtspsrc.c:
rtspsrc: Actually use the receive lock when receiving, not the send
lock
Download
========
https://download.gnome.org/sources/gst-plugins-good/1.12/gst-plugins-good-1.12.2.tar.xz
(3.33M)
sha256sum: 5591ee7208ab30289a30658a82b76bf87169c927572d9b794f3a41ed48e1ee96
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