ChangeLog
=========
2017-01-30 Sebastian Dröge <sl...@coaxion.net>
* configure.ac:
releasing 1.10.3
2017-01-30 13:30:51 +0200 Sebastian Dröge <sebast...@centricular.com>
* po/fr.po:
* po/nb.po:
* po/sr.po:
po: Update translations
2017-01-30 12:35:04 +0200 Sebastian Dröge <sebast...@centricular.com>
* gst-libs/gst/audio/audio-resampler-x86-sse41.c:
audio-resampler: Fix integer overflow in clamping code
https://bugzilla.gnome.org/show_bug.cgi?id=777921
2017-01-20 12:41:16 +0200 Sebastian Dröge <sebast...@centricular.com>
* gst-libs/gst/riff/riff-media.c:
riff-media: Don't divide block align by zero channels
https://bugzilla.gnome.org/show_bug.cgi?id=777525
2017-01-20 08:02:38 +0200 Sebastian Dröge <sebast...@centricular.com>
* gst/subparse/samiparse.c:
samiparse: Check that the string has a non-zero length before
overwriting the last byte with '\0'
https://bugzilla.gnome.org/show_bug.cgi?id=777502
2017-01-15 18:42:34 +0100 Sebastian Dröge <sebast...@centricular.com>
* gst-libs/gst/riff/riff-media.c:
riff-media: Don't recurse in for nested WAVEFORMATEX
There was already a check for that, but it failed because
subformat_guid[0] is a guint32 and that is then casted implicitely to
a
guint16 when recursing... just that we checked the uncasted value.
This caused an infinite recursion and thus stack overflow.
https://bugzilla.gnome.org/show_bug.cgi?id=777265
2017-01-15 18:31:56 +0100 Sebastian Dröge <sebast...@centricular.com>
* gst-libs/gst/riff/riff-media.c:
riff-media: Check for valid channels/rate before using the values
Otherwise we might divide by zero or otherwise create invalid caps.
https://bugzilla.gnome.org/show_bug.cgi?id=777262
2017-01-11 18:24:38 +0200 Sebastian Dröge <sebast...@centricular.com>
* gst-libs/gst/video/video-converter.c:
video-converter: Fix crashes in fast-paths when converting interlaced
formats with different vertical subsampling
E.g. the following pipelines fail because chroma values after the last
line are read (note: 486 % 4 == 2):
gst-launch-1.0 videotestsrc !
"video/x-raw,interlace-mode=interleaved,width=720,height=486,format=UYVY" !
videoconvert ! "video/x-raw,format=I420" ! fakesink
gst-launch-1.0 videotestsrc !
"video/x-raw,interlace-mode=interleaved,width=720,height=486,format=I420" !
videoconvert ! "video/x-raw,format=UYVY" ! fakesink
gst-launch-1.0 videotestsrc !
"video/x-raw,interlace-mode=interleaved,width=720,height=486,format=I420" !
videoconvert ! "video/x-raw,format=AYUV" ! fakesink
2017-01-10 08:57:51 -0300 Thibault Saunier <thibault.saun...@osg.samsung.com>
* gst-libs/gst/pbutils/encoding-profile.c:
pbutils: Fix annotation in gst_encoding_profile_set_preset
2017-01-09 21:25:26 +1100 Jan Schmidt <j...@centricular.com>
* gst-libs/gst/video/video.c:
gst_video_guess_framerate: Don't throw away all precision
When operating on framerates near 10000fps, at least keep 1
digit of precision for calculations
2017-01-04 11:21:51 -0300 Thibault Saunier <thibault.saun...@osg.samsung.com>
* gst/encoding/gstencodebin.c:
encodebin: Fix stream_group_free when creating it went bad
Avoiding trying to use NULL pointers
2016-12-30 17:55:18 +0100 Mark Nauwelaerts <mn...@users.sourceforge.net>
* gst/playback/gstplaysink.c:
playsink: do not link to audio or video filter using padname
... as a sinkpad need not be called "sink", and it is not the case
for e.g. timeoverlay (and friends).
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=776623
2017-01-02 12:54:32 +0000 Tim-Philipp Müller <t...@centricular.com>
* gst/encoding/gstencodebin.c:
encodebin: fix queue property types when setting
2015-11-25 11:30:42 +0000 Stuart Weaver <stuart.wea...@datapath.co.uk>
* gst-libs/gst/rtsp/gstrtspurl.c:
rtsp-url: unescape special chars in user/pass part of URL
This way special characters such as '@' can be used in
usernames or passwords, e.g.
rtsp://view:%40dm%4An@<IP-ADDR>/media/camera1
will now parse username and password into:
User: view
Pass: @dm:n
https://bugzilla.gnome.org/show_bug.cgi?id=758389
2016-09-02 15:23:18 +0200 Carlos Rafael Giani <d...@pseudoterminal.org>
* gst/audiotestsrc/gstaudiotestsrc.c:
audiotestsrc: Fix incorrect start of tick waveform
Make sure ticks start with an accumulator value of 0 by incrementing
it
after filling in samples instead of before and by resetting the
accumulator
every time a tick begins. This prevents it from being discontinuous
at the
beginning of the tick.
https://bugzilla.gnome.org/show_bug.cgi?id=774050
2016-12-22 18:47:19 +0100 Nicolas Dechesne <nicolas.deche...@linaro.org>
* tools/gst-play.c:
tools: gst-play: set GST_GL_XINITHREADS
This ensure that XInitThreads is called and so gl contexts are
properly
initialized.
https://bugzilla.gnome.org/show_bug.cgi?id=776403
2016-12-21 00:11:06 +1100 Jan Schmidt <j...@centricular.com>
* gst/playback/gstparsebin.c:
parsebin: Ignore failure to send sticky events
When plugging and then exposing a parser, don't fail
if it fails to send sticky events. The most likely
reason is that things were flushed due to the app
immediately doing a seek, but we can't detect flushing
separately to other error conditions without a
gst_pad_send_event_full() core function that returns
a GstFlowReturn.
2016-12-15 16:29:02 +0200 Sebastian Dröge <sebast...@centricular.com>
* gst/playback/gstdecodebin2.c:
decodebin: For adaptive streaming, ensure to put the buffering
multiqueue after a parser or demuxer
There are cases when there is no demuxer involved that could do the
buffering, e.g. HLS with raw MP3 or AAC. In this case we want to place
the buffering multiqueue after the parser.
Before this change, we've considered the first element after the
adaptive streaming demuxer as a parser. This is not always true, e.g.
id3demux. Instead we now wait until we actually have a parser (or
decoder).
Fixes playback on such HLS streams.
2016-12-09 17:36:47 +0200 Sebastian Dröge <sebast...@centricular.com>
* gst-libs/gst/tag/gstxmptag.c:
xmptag: Don't leak the namespace string if there are multiple
https://bugzilla.gnome.org/show_bug.cgi?id=775887
2016-12-09 17:57:52 +1100 Jan Schmidt <j...@centricular.com>
* gst-libs/gst/tag/id3v2.c:
id3v2: Add missing overrun check for frame sizes
When frames claim to have a footer, ensure they
are large enough to contain one to avoid an invalid
read overrun.
Spotted by Joshua Yabut
2016-12-06 16:29:23 +0200 Sebastian Dröge <sebast...@centricular.com>
* gst-libs/gst/tag/gsttagdemux.c:
tagdemux: Fix crash when shutting down element during getrange()
Ensure that nothing is in any of the streaming thread functions
anymore when going from PAUSED to READY. While the parent's state
change
function has deactivated all pads, there is nothing preventing
downstream from activating our srcpad again and calling the getrange()
function. Although we're in READY!
https://bugzilla.gnome.org/show_bug.cgi?id=775687
2016-11-04 16:41:05 +0000 Vincent Penquerc'h
<vincent.penque...@collabora.co.uk>
* ext/opus/gstopusdec.c:
opusdec: fix 120 ms buffers being wrongly emitted
Using the max 120 ms buffer size to ensure we have enough space
for decoded data meant that Opus could actually return 120 ms'
worth of data.
https://bugzilla.gnome.org/show_bug.cgi?id=771723
2016-09-26 10:50:52 +0100 Vincent Penquerc'h
<vincent.penque...@collabora.co.uk>
* ext/opus/gstopusdec.c:
opusdec: fix "buffer too small" error
Always supply a buffer with max size to the decoder, as we
can't really decide how many samples will be in the lost packet
based on the timestamps we get.
https://bugzilla.gnome.org/show_bug.cgi?id=771723
2016-10-06 11:44:11 +0100 Vincent Penquerc'h
<vincent.penque...@collabora.co.uk>
* ext/opus/gstopusdec.c:
opusdec: interpret zero duration as unknown
This fixes missing audio when we get buffers with zero
duration, denoting unknown duration. When several such
buffers are received in a row, they're all at the same
timestamp, with zero duration.
https://bugzilla.gnome.org/show_bug.cgi?id=771723
2016-11-29 16:26:22 +0100 Jan Alexander Steffens (heftig)
<jan.steff...@gmail.com>
* tests/check/elements/multifdsink.c:
multifdsink: Add a test involving a slow client
https://bugzilla.gnome.org/show_bug.cgi?id=774908
2016-11-23 14:35:04 +0100 Jan Alexander Steffens (heftig)
<jan.steff...@gmail.com>
* gst/tcp/gstmultihandlesink.c:
multihandlesink: Update bufpos in a separate pass
If a client gets dropped and the iteration gets restarted, bufpos is
incremented again for all clients that preceded the dropped one,
causing
havoc.
Adjust the bufpos for all clients first before trying to drop any.
https://bugzilla.gnome.org/show_bug.cgi?id=774908
2016-11-29 15:30:43 +0100 Jan Alexander Steffens (heftig)
<jan.steff...@gmail.com>
* gst/tcp/gstmultihandlesink.c:
multihandlesink: Fix buffers-queued being off by one
max_buffer_usage is the index of the oldest buffer in the queue,
starting at zero, not the number of buffers queued.
find_limits returns the index of the oldest buffer that satisfies the
limits in its min_idx parameter, not the number of buffers needed. Fix
this use too in order to keep passing the tests that read
buffers-queued.
https://bugzilla.gnome.org/show_bug.cgi?id=775351
2016-12-01 15:12:59 +0200 Sebastian Dröge <sebast...@centricular.com>
* ext/ogg/gstoggdemux.c:
oggdemux: Don't end up ignoring caps just because there are no
headers for this stream
https://bugzilla.gnome.org/show_bug.cgi?id=775459
2016-12-01 19:57:47 +0200 Sebastian Dröge <sebast...@centricular.com>
* gst/subparse/gstssaparse.c:
ssaparse: Free initialization section before storing the next one
If getting multiple caps events.
https://bugzilla.gnome.org/show_bug.cgi?id=775480
Download
========
https://download.gnome.org/sources/gst-plugins-base/1.10/gst-plugins-base-1.10.3.tar.xz
(2.92M)
sha256sum: e6299617d705a0cbfb535107c1d3a8fc0f0967f14193a8c5c7583f46a88b1710
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