Commit af4c3ad0 authored by Sebastian Dröge's avatar Sebastian Dröge 🍵

Release 1.10.3

parent 9a410b2d
=== release 1.10.3 ===
2017-01-30 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.10.3
2017-01-30 13:30:51 +0200 Sebastian Dröge <sebastian@centricular.com>
* po/fr.po:
* po/nb.po:
* po/sr.po:
po: Update translations
2017-01-30 12:35:04 +0200 Sebastian Dröge <sebastian@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 <sebastian@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 <sebastian@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 <sebastian@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 <sebastian@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 <sebastian@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.saunier@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 <jan@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.saunier@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 <mnauw@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 <tim@centricular.com>
* gst/encoding/gstencodebin.c:
encodebin: fix queue property types when setting
2015-11-25 11:30:42 +0000 Stuart Weaver <stuart.weaver@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 <dv@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.dechesne@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 <jan@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 <sebastian@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 <sebastian@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 <jan@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 <sebastian@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.penquerch@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.penquerch@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.penquerch@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.steffens@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.steffens@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.steffens@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 <sebastian@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 <sebastian@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
=== release 1.10.2 ===
2016-11-29 Sebastian Dröge <slomo@coaxion.net>
2016-11-29 16:20:54 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
releasing 1.10.2
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
* gst-plugins-base.doap:
* win32/common/_stdint.h:
* win32/common/config.h:
Release 1.10.2
2016-11-29 15:28:59 +0200 Sebastian Dröge <sebastian@centricular.com>
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/el.po:
* po/en_GB.po:
* po/eo.po:
* po/es.po:
* po/eu.po:
* po/fi.po:
* po/fr.po:
* po/gl.po:
* po/hr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/lv.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ro.po:
* po/ru.po:
* po/sk.po:
* po/sl.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/tr.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
Update .po files
2016-11-29 14:09:18 +0200 Sebastian Dröge <sebastian@centricular.com>
# GStreamer 1.10 Release Notes
GStreamer 1.10.0 was originally released on 1st November 2016.
The latest bug-fix release in the 1.10 series is [1.10.2](#1.10.2) and was
released on 29 November 2016.
The latest bug-fix release in the 1.10 series is [1.10.3](#1.10.3) and was
released on 30 January 2017.
The GStreamer team is proud to announce a new major feature release in the
stable 1.x API series of your favourite cross-platform multimedia framework!
......@@ -13,7 +13,7 @@ improvements.
See [https://gstreamer.freedesktop.org/releases/1.10/][latest] for the latest
version of this document.
*Last updated: Tuesday 29 Nov 2016, 12:30 UTC [(log)][gitlog]*
*Last updated: Monday 30 Jan 2017, 12:00 UTC [(log)][gitlog]*
[latest]: https://gstreamer.freedesktop.org/releases/1.10/
[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.10/release-notes-1.10.md
......@@ -1103,7 +1103,7 @@ GIT logs or ChangeLogs of the particular modules.
### 1.10.2
The first 1.10 bug-fix release (1.10.2) was released on 29 November 2016.
The second 1.10 bug-fix release (1.10.2) was released on 29 November 2016.
This release only contains bugfixes and it should be safe to update from 1.10.x.
#### Major bugfixes in 1.10.2
......@@ -1111,7 +1111,9 @@ This release only contains bugfixes and it should be safe to update from 1.10.x.
- Security-relevant bugfix in the FLI/FLX/FLC decoder (CVE-2016-9634,
CVE-2016-9635, CVE-2016-9636)
- Various fixes for crashes, assertions and other failures on fuzzed input
files (among others, thanks to Hanno Böck for testing and reporting)
files. Among others, thanks to Hanno Böck for testing and reporting
(CVE-2016-9807, CVE-2016-9808, CVE-2016-9809, CVE-2016-9810, CVE-2016-9811,
CVE-2016-9812, CVE-2016-9813).
- SAVP/SAVPF profile in gst-rtsp-server works for live streams again, and the
correct MIKEY policy message is generated
- Further OpenGL related bugfixes
......@@ -1124,6 +1126,32 @@ GIT logs or ChangeLogs of the particular modules.
[buglist-1.10.2]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=168172&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.10.2
<a name="1.10.3"></a>
### 1.10.3
The third 1.10 bug-fix release (1.10.3) was released on 30 January 2017.
This release only contains bugfixes and it should be safe to update from 1.10.x.
#### Major bugfixes in 1.10.3
- Various fixes for crashes, assertions, deadlocks and memory leaks on fuzzed
input files and in other situations
- Regression fixes for souphttpsrc with redirection tracking and retrying
- Regression fix for gst-rtsp-server not handling TCP-only medias anymore
- Various other bugfixes the RTP/RTSP codebase
- vp8enc works again on 32 bit Windows
- Fixes to Opus PLC handling in the decoder
- Fix for stream corruption in multihandlesink when removing clients
- gst-libav was updated to ffmpeg 3.2.2
- ... and many, many more!
For a full list of bugfixes see [Bugzilla][buglist-1.10.3]. Note that this is
not the full list of changes. For the full list of changes please refer to the
GIT logs or ChangeLogs of the particular modules.
[buglist-1.10.3]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=187054&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.10.3
## Known Issues
- iOS builds with iOS 6 SDK and old C++ STL. You need to select iOS 6 instead
......@@ -1134,8 +1162,6 @@ GIT logs or ChangeLogs of the particular modules.
- Building applications with Android NDK r13 on Windows does not work. Other
platforms and earlier/later versions of the NDK are not affected.
[Bug #772842](https://bugzilla.gnome.org/show_bug.cgi?id=772842)
- vp8enc crashes on 32 bit Windows, but was working fine in 1.6. 64 bit Windows is unaffected.
[Bug #763663](https://bugzilla.gnome.org/show_bug.cgi?id=763663)
## Schedule for 1.12
......@@ -1144,9 +1170,9 @@ development version leading up to the stable 1.12 release. The development
of 1.11/1.12 will happen in the git master branch.
The plan for the 1.12 development cycle is yet to be confirmed, but it is
expected that feature freeze will be around early/mid-January,
expected that feature freeze will be around early/mid-February,
followed by several 1.11 pre-releases and the new 1.12 stable release
in March.
in April.
1.12 will be backwards-compatible to the stable 1.10, 1.8, 1.6, 1.4, 1.2 and
1.0 release series.
......
Release notes for GStreamer Base Plugins 1.10.2
Release notes for GStreamer Base Plugins 1.10.3
The GStreamer team is proud to announce the second bugfix release in the stable
The GStreamer team is proud to announce the third bugfix release in the stable
1.10 release series of your favourite cross-platform multimedia framework!
......@@ -58,10 +58,21 @@ contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Bugs fixed in this release
* 774911 : mikey: gst_mikey_message_new_from_caps does not generate a correct message
* 774585 : videotimecode: Fix incorrect nsec_since_daily_jam calculation
* 774902 : typefind: out of bounds memory read in windows_icon_typefind
* 775224 : subtitleoverlay: Caps memory leak when failing to get sinkpad from subtitle renderer
* 758389 : rtsp-url: unescape special chars in user/pass part of URL
* 771723 : opusdec: too short buffers trigger error instead of PLC
* 774908 : multifdsink removing slow client corrupts matroska stream
* 775351 : multihandlesink: buffers-queued is off by one
* 775459 : oggdemux: Hangs on divx-bandits-sample file
* 775480 : ssaparse: memory leak in gst_ssa_parse_setcaps
* 775687 : tagdemux: Crash if getrange/shutdown happen at the same time
* 775887 : qtdemux/xmptag: memory leak in gst_tag_list_from_xmp_buffer
* 776403 : gst-play should set GST_GL_XINITHREADS so that XinitThread() is called appropriately
* 776623 : playbin/playsink fail to use timeoverlay as video-filter
* 777262 : riff-media: floating point exception in gst_riff_create_audio_caps
* 777265 : riff: stack overflow in gst_riff_create_audio_caps
* 777502 : samiparse: heap oob in html_context_handle_element
* 777525 : floating point exception in gst_riff_create_audio_caps (different than #777262)
* 777921 : audio-resampler: integer overflow in clamping code
==== Download ====
......@@ -98,11 +109,14 @@ subscribe to the gstreamer-devel list.
Contributors to this release
* Edward Hervey
* Garima Gaur
* Matthew Waters
* Carlos Rafael Giani
* Jan Alexander Steffens (heftig)
* Jan Schmidt
* Mark Nauwelaerts
* Nicolas Dechesne
* Sebastian Dröge
* Stuart Weaver
* Thibault Saunier
* Tim-Philipp Müller
* Ulf Olsson
* Vivia Nikolaidou
* Vincent Penquerc'h
 
\ No newline at end of file
......@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
dnl initialize autoconf
dnl releases only do -Wall, git and prerelease does -Werror too
dnl use a three digit version number for releases, and four for git/prerelease
AC_INIT([GStreamer Base Plug-ins],[1.10.2],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
AC_INIT([GStreamer Base Plug-ins],[1.10.3],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
AG_GST_INIT
......@@ -56,7 +56,7 @@ dnl 1.2.5 => 205
dnl 1.10.9 (who knows) => 1009
dnl
dnl sets GST_LT_LDFLAGS
AS_LIBTOOL(GST, 1002, 0, 1002)
AS_LIBTOOL(GST, 1003, 0, 1003)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.10.0
......
......@@ -3,7 +3,7 @@
<description>Adds multiple streams</description>
<filename>../../gst/adder/.libs/libgstadder.so</filename>
<basename>libgstadder.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>ALSA plugin library</description>
<filename>../../ext/alsa/.libs/libgstalsa.so</filename>
<basename>libgstalsa.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Elements used to communicate with applications</description>
<filename>../../gst/app/.libs/libgstapp.so</filename>
<basename>libgstapp.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Convert audio to different formats</description>
<filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
<basename>libgstaudioconvert.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Adjusts audio frames</description>
<filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename>
<basename>libgstaudiorate.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Resamples audio</description>
<filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename>
<basename>libgstaudioresample.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Creates audio test signals of given frequency and volume</description>
<filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename>
<basename>libgstaudiotestsrc.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Read audio from CD in paranoid mode</description>
<filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename>
<basename>libgstcdparanoia.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>various encoding-related elements</description>
<filename>../../gst/encoding/.libs/libgstencodebin.so</filename>
<basename>libgstencodebin.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>GIO elements</description>
<filename>../../gst/gio/.libs/libgstgio.so</filename>
<basename>libgstgio.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>libvisual visualization plugins</description>
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
<basename>libgstlibvisual.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>ogg stream manipulation (info about ogg: http://xiph.org)</description>
<filename>../../ext/ogg/.libs/libgstogg.so</filename>
<basename>libgstogg.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>OPUS plugin library</description>
<filename>../../ext/opus/.libs/libgstopus.so</filename>
<basename>libgstopus.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>Pango-based text rendering and overlay</description>
<filename>../../ext/pango/.libs/libgstpango.so</filename>
<basename>libgstpango.so</basename>
<version>1.10.2</version>
<version>1.10.3</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>various playback elements</description>