Commit 19174006 authored by Sebastian Dröge's avatar Sebastian Dröge

Release 1.3.2

parent ab756a68
=== release 1.3.2 ===
2014-05-21 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.3.2
2014-05-21 10:50:56 +0200 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From 211fa5f to 1f5d3c3
2014-05-21 10:43:49 +0200 Sebastian Dröge <sebastian@centricular.com>
* tests/check/libs/video.c:
video: And check comparison for real
2014-05-21 10:40:32 +0200 Sebastian Dröge <sebastian@centricular.com>
* tests/check/libs/video.c:
video: Fix broken comparison in unit test
libs/video.c:540:50: error: comparison of constant 2 with boolean expression is always false
[-Werror,-Wtautological-constant-out-of-range-compare]
&& !GST_VIDEO_INFO_N_PLANES (&vinfo) > 2) {
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ^ ~
2014-05-20 15:59:53 +0200 Wim Taymans <wtaymans@redhat.com>
* gst-libs/gst/rtsp/gstrtsptransport.h:
rtsp-transport: clarify port usage
Comment in the docs what the client_port and server_port fields are used
for in TCP mode (if the application wants to set those values).
2014-05-20 11:18:56 +0200 Michael Olbrich <m.olbrich@pengutronix.de>
* gst-libs/gst/allocators/gstdmabuf.c:
dmabuf: share the mapping with shared copies of the memory
With lots of shared memory instances (e.g. created by a RTP payloader) the
overhead of duplicating the file descriptor and creating extra mappings is
significant. To avoid this, the parent memory maps the whole region and the
shared copies just reuse the same mapping.
https://bugzilla.gnome.org/show_bug.cgi?id=730441
2014-05-19 13:28:52 +0200 Göran Jönsson <goranjn@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: Add read source on write socket.
Add a read source on write socket when lost tunnel.
To be able to detect when clint closes get channel.
This is already done in gst_rtsp_source_dispatch_write but
only when the queue is empty.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730368
2014-05-20 09:48:56 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/playback/gstplaysink.c:
playsink: Always take the playsink lock when adding or removing pad probes
Otherwise we might end up inside the callback without having stored
the probe id... then try to remove that probe (not!) from the callback
and wait forever for the pad to unblock.
2014-05-19 13:57:41 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
* ext/alsa/gstalsasink.c:
alsasink: pass correct error to g_strerror
The error we get is a negated errno.
While there, fix a couple typos in messages.
2014-05-19 11:17:33 +0200 Sebastian Dröge <sebastian@centricular.com>
* tools/gst-play.c:
gst-play: Free playlist_file string if only printing the version
2014-05-13 14:08:20 +0600 Anuj Jaiswal <anuj.jaiswal@samsung.com>
* tools/gst-play.c:
audio_sink and video_sink leakage fixed
https://bugzilla.gnome.org/show_bug.cgi?id=730010
2014-05-13 11:51:55 +0200 Edward Hervey <edward@collabora.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: Don't use argument for local storage
By re-using the uri argument for storing local data, we could end up in
a situation where we would free uri ... which would actually be the
string passed in argument.
Instead explicitely use a local variable. Fixes double-free issues.
CID #1212176
2014-05-12 13:18:50 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst-libs/gst/video/video-info.c:
video-info: Also check the stride and offset are equal
gst_video_info_is_equal() was not checking if stride and offset
had changed.
https://bugzilla.gnome.org/show_bug.cgi?id=729896
2014-05-12 17:17:07 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/video/gstvideodecoder.c:
videodecoder: Free data after removing it from the list
While it wouldn't have caused any failures (g_list_remove doesn't dereference
the provided pointer), it does make the code cleaner.
CID #1212174
2014-05-12 17:15:17 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/sdp/gstmikey.c:
mikey: Actually replace payload ...
This function is intented to replace the payload, let's actually do that
instead of putting back the same (freed) payload
CID #1212175
2014-05-12 17:13:50 +0200 Edward Hervey <bilboed@bilboed.com>
* gst-libs/gst/sdp/gstmikey.c:
mikey: Free MikeyPayload in error cases
CID #1212135
CID #1212136
CID #1212137
CID #1212138
2014-05-10 23:50:44 +0200 Thibault Saunier <tsaunier@gnome.org>
* ext/pango/gstbasetextoverlay.c:
pango: Do not try to add a feature to a caps features ANY
It does not makes sense and asserts
2014-05-09 15:32:18 +0100 Tim-Philipp Müller <tim@centricular.com>
* gst-libs/gst/tag/gstxmptag.c:
tag: xmp: fix leaks in error code paths
CID 1212133
2014-05-06 11:12:19 +0200 Göran Jönsson <goranjn@axis.com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
rtspconnection: Reset control_stream.
Reset control_stream when gst_rtsp_connection_close.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729632
2014-04-15 14:51:46 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
* gst-libs/gst/video/gstvideodecoder.c:
videodecoder: Retry setting configuration with modified config
Buffer pool set_config() may return FALSE if requested configuration needed small
changes. Reget the config and try setting it again. This ensure we have a configured
pool if possible.
2014-05-08 17:10:26 +0200 Wim Taymans <wtaymans@redhat.com>
* gst/playback/gsturidecodebin.c:
uridecodebin: use downloadbuffer for download buffering
Use the new downloadbuffer element to implement the download buffering
feature
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680183
2014-05-06 13:01:32 -0400 Luis de Bethencourt <luis@debethencourt.com>
* ext/ogg/gstoggmux.c:
oggmux: push eos event when empty pad data
If gst_ogg_mux_queue_pads returns NULL it means we are at EOS, because we get a
NULL buffer and this function never sets bestpad.
https://bugzilla.gnome.org/show_bug.cgi?id=729315
2014-05-06 08:07:38 +0000 Руслан Ижбулатов <lrn1986@gmail.com>
* configure.ac:
configure: Use X11 detection macro from common
https://bugzilla.gnome.org/show_bug.cgi?id=729621
2014-05-06 07:51:11 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/examples/playback/playback-test.c:
examples: playback-test: fix crashes when setting buffer-size
playbin's buffer-size property takes a gint, not a gint64,
so only pass the bits expected to the vararg function, or
the terminator might not be found, leading to crashes, esp.
with negative numbers.
Spotted by Ravi Kiran K N <ravi.kiran@samsung.com>
https://bugzilla.gnome.org/show_bug.cgi?id=729617
2014-05-06 07:50:16 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/examples/playback/playback-test.c:
examples: fix indentation of playback-test
2014-05-06 08:13:24 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/examples/playback/playback-test.c:
Revert "playback-test: Set buffer-size only for non-negative size"
This reverts commit 07a637e2847d56d0f2b0c0ac9095bf37dd324e26.
2014-05-06 11:31:18 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
* tests/examples/playback/playback-test.c:
playback-test: Set buffer-size only for non-negative size
https://bugzilla.gnome.org/show_bug.cgi?id=729617
2014-05-05 23:29:44 -0400 Luis de Bethencourt <luis@debethencourt.com>
* win32/common/libgstpbutils.def:
win32: Update defs file
commit 622007e7db7e3d32bf8e04e673e057897b646220 added the function
gst_discoverer_info_get_missing_elements_installer_details (). It needs to be
added to the defs file.
2014-05-04 15:54:54 +0000 Руслан Ижбулатов <lrn1986@gmail.com>
* configure.ac:
* gst-libs/gst/rtsp/Makefile.am:
rtsp: Link to ws2_32 on Windows
Needed for getsockname and setsockopt
https://bugzilla.gnome.org/show_bug.cgi?id=729514
2014-05-04 15:54:06 +0000 Руслан Ижбулатов <lrn1986@gmail.com>
* configure.ac:
Make X11 detection more precise
Don't be content with just X11/Xlib.h, check for X11/XKBlib.h as well.
This prevents false positives (for example, from partial X11 headers
installed by tcl/tk).
https://bugzilla.gnome.org/show_bug.cgi?id=729513
2014-05-04 15:57:35 +0000 Руслан Ижбулатов <lrn1986@gmail.com>
* tests/examples/playback/playback-test.c:
tests: fix printf format compiler warning in playback test on win32
https://bugzilla.gnome.org/show_bug.cgi?id=729515
2014-05-04 18:14:54 +0100 Tim-Philipp Müller <tim@centricular.com>
* tests/check/libs/.gitignore:
Add new unit test binary to .gitignore
2014-01-14 15:39:55 +0100 Thibault Saunier <thibault.saunier@collabora.com>
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/gstdiscoverer-types.c:
* gst-libs/gst/pbutils/gstdiscoverer.c:
* gst-libs/gst/pbutils/gstdiscoverer.h:
* gst-libs/gst/pbutils/pbutils-private.h:
* tools/gst-discoverer.c:
discoverer: Add APIs to simply get installer details for missing plugins
Currently the API is far from optimal and the user has to work around
our badly defined API to simply install missing plugins.
API:
new:
gst_discoverer_info_get_missing_elements_installer_details
deprecated:
gst_discoverer_info_get_misc
gst_discoverer_stream_info_get_misc
https://bugzilla.gnome.org/show_bug.cgi?id=720596
2014-05-03 20:48:27 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
2014-05-03 18:57:38 +0200 Sebastian Dröge <sebastian@centricular.com>
* tests/check/Makefile.am:
textoverlay: Link unit test with the local version of the library, not an installed one
=== release 1.3.1 ===
2014-05-03 Sebastian Dröge <slomo@coaxion.net>
2014-05-03 17:50:10 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
releasing 1.3.1
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* 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-ivorbisdec.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.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-libs/gst/audio/gstaudiopack-dist.c:
* gst-libs/gst/video/video-orc-dist.c:
* gst-plugins-base.doap:
* gst/adder/gstadderorc-dist.c:
* gst/audioconvert/gstaudioconvertorc-dist.c:
* gst/videoconvert/gstvideoconvertorc-dist.c:
* gst/videoscale/gstvideoscaleorc-dist.c:
* gst/videotestsrc/gstvideotestsrcorc-dist.c:
* gst/volume/gstvolumeorc-dist.c:
* win32/common/_stdint.h:
* win32/common/config.h:
* win32/common/gstrtsp-enumtypes.c:
* win32/common/video-enumtypes.c:
* win32/common/video-enumtypes.h:
Release 1.3.1
2014-05-03 17:48:04 +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
2014-05-03 17:22:10 +0200 Sebastian Dröge <sebastian@centricular.com>
This is GStreamer Base Plugins 1.3.1
This is GStreamer Base Plugins 1.3.2
Changes since 1.2:
......@@ -45,6 +45,8 @@ New API:
events and merge custom tags into them consistently.
• playbin/playsink has support for application provided audio and video
filters.
• GstDiscoverer has new and simplified API to get details about missing
plugins and information to pass to the plugin installer.
• The GL library was merged from gst-plugins-gl to gst-plugins-bad,
providing a generic infrastructure for handling GL inside GStreamer
pipelines and a plugin with some elements using these, especially
......@@ -62,6 +64,14 @@ Major changes:
of the existing V4L2 elements and the corresponding
infrastructure.
The v4l2videodec element replaces the mfcdec element.
∘ New downloadbuffer element that replaces the download
buffering feature of queue2. Compared to queue2's code
it is much simpler and only for this single use case.
A noteworthy new feature is that it's downloading gaps
in the already downloaded stream parts when nothing else
is to be downloaded.
This is now used by playbin when download buffering is
enabled.
∘ rtpstreampay and rtpstreamdepay elements for transmitting
RTP packets over a stream API (e.g. TCP) according to
RFC 4571.
......@@ -78,7 +88,7 @@ Major changes:
are available on OS X and iOS now.
• Other changes:
∘ gst-libav now uses libav 10, and gained support for H265/HEVC.
∘ gst-libav now uses libav 10.1, and gained support for H265/HEVC.
∘ Support for hardware codecs and special memory types has been
improved with bugfixes and feature additions in various plugins
and base classes.
......@@ -95,6 +105,9 @@ Major changes:
reliable now and supports more HLS features like trick modes.
Also fragments are pushed downstream while they're downloaded
now instead of waiting for each fragment to finish.
∘ dashdemux and mssdemux are now also pushing fragments downstream
while they're downloaded instead of waiting for each fragment to
finish.
∘ videoflip can automatically flip based on the orientation tag.
∘ openjpeg supports the OpenJPEG2 API.
∘ gst-rtsp-server supports SRTP and MIKEY now.
......@@ -107,4 +120,3 @@ Things to look out for:
element.
• The mfcdec element was removed and replaced by v4l2videodec.
• osxvideosink is only available in OS X 10.6 or newer.
Release notes for GStreamer Base Plugins 1.3.1
Release notes for GStreamer Base Plugins 1.3.2
The GStreamer team is pleased to announce the first release of the unstable
The GStreamer team is pleased to announce the second release of the unstable
1.3 release series. The 1.3 release series is adding new features on top of
the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release
series of the GStreamer multimedia framework. The unstable 1.3 release series
......@@ -66,80 +67,21 @@ contains a set of less supported plugins that haven't passed the
gst-libav
contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Bugs fixed in this release
* 684030 : typefinding: mp4 with video and dts ES detected as DTS audio
* 725078 : audiobasesink: clip start samples to match clipped timestamp from skew algorithm
* 708633 : adder: Should not take channel mask in consideration when in mono or stereo
* 540941 : v4l2: RGB32 should be mapped to xRGB instead of RGBx
* 646577 : rtppayload: Make RTP time information accessible
* 670690 : audioresample: missing configure checks for SSE / SSE2
* 678402 : Device discovery/listing replacement for GstPropertyProbe
* 678590 : subparse: Add support for LRC subtitles
* 679031 : playbin/playsink: Add support for audio and video filters
* 687183 : videodecoder: Allow to negotiate a buffer pool before output format is known
* 702230 : audioringbuffer: Don't access timestamps array if not acquired
* 707361 : video: Add support for 64x32 tiled NV12 color format
* 707636 : dashdemux: offline playback not buffering correctly
* 708680 : typefind: Add typefind function for H265
* 708921 : pbutils: Add codec-utility functions to support h265
* 708991 : audiocdsrc: invalid musicbrainz discids because of trailing data tracks
* 709588 : encodebin: Handle changes in encoding_profile::restriction during playback
* 709646 : videotestsrc: Could implement duration query when num-buffers is set
* 709755 : alsa: add channel map API support
* 709814 : [examples/overlay] avoid to unref sink if not found. Also fix logic to find a sink in one of the example.
* 709858 : theoraenc: Do nothing when flushing the encoder when no caps were set
* 710760 : videoconvert: remove unneeded guint comparison
* 711094 : videodecoder: improve max-error handling
* 711258 : sdp: fix duplicate 'const' declaration warnings
* 712798 : videometa: add GstVideoGLTextureUploadMeta buffer pool option
* 719383 : rtpbasepayload: Perfect timestamps confusingly explained
* 719415 : rtpbasepayload: Expose running time of last processed buffer
* 719850 : convertframe: remove trivial memory leak
* 719890 : videodecoder: Add API to get the currently pending, parsed frame size
* 720103 : videodecoder: Introduce sink_query/src_query
* 720124 : tests/examples/overlay/qt-videooverlay.cpp has incorrect include from Qt
* 720162 : tests: Add test for rtpbasepayload/-depayload
* 720205 : playback: add video/x-raw(ANY) to default raw caps
* 720215 : sdp: parse encryption key field
* 720219 : rtsptransport: allow getting mime type by profile
* 720389 : videodecoder: should release buffer pool sooner
* 720810 : audio/video: Initialize all {audio|video}info fields
* 720999 : Missing annotation for GstColorBalance interface
* 721103 : test-effect-switch errors out with not-negotiated after a while
* 721701 : videoconvert: I420 to BGRA conversion is slower than in 0.10
* 721953 : pango: basetextoverlay: handle video/x-raw(ANY) if downstream supports the GstVideoOverlayCompositionMeta API
* 722330 : streamsplitter: negotiation problems with parsers
* 722491 : playbin: remove duplicate assignment
* 722682 : oggmux: problems with vp8 stream
* 723096 : decodebin: Make it possible to register multiple handlers to decodebin's autoplug-select signal
* 723271 : videotestsrc: fix a warning if downstream does not propose a buffer pool
* 723328 : gstrtpbase(|de)payload: add more unit tests and fix bugs
* 723492 : gst-plugins-base: Do not build check tests for disabled plugins
* 723507 : jsseek: Add missing HAVE_X check
* 724393 : rtspconnection: allow specifying an anchor certificate database
* 724509 : audioconvert: outputs silence when converting certain mono caps to certain other mono caps
* 724828 : playbin: improve autoplug_query_caps return
* 724893 : playsinkconvertbin: improve gst_play_sink_convert_bin_getcaps return
* 725034 : all plugin sets but -base don't install gtk-doc docs without '--enable-gtk-doc'
* 725206 : rtspconnection: Missing include file
* 725479 : gst-plugins-base: Ignore gcov intermediate files
* 725521 : docs: Fix argument and annotation typos, add missing annotations and remove duplicate section
* 725658 : Removing some GnomeVFS left bits
* 725837 : pango: textoverlay: lot of warnings in debug log with framerate=0/1
* 725878 : rtspconnection: headers in GET response not configurable for tunnels
* 725898 : Lose data when producing data faster than sendt during tunneling rtps/rtp(TCP)
* 726433 : rtspconnection: setsockopt() argument 4 is not properly casted for W32
* 726641 : rtspconnection: connection_poll() not working correctly
* 727498 : videodecoder: deactivates downstream bufferpool
* 728772 : rtspconnection: stuck in teardown
* 728845 : gst-play: add option to supply input media-files from a playlist file
* 728907 : rtspconnection: add more tests
* 729114 : audiodecoder: default caps nego will manually fixate non-mutable caps
* 729117 : rtpbuffer: fix memory leak when gst_rtp_buffer_map fails
* 729195 : videotestsrc: undefined behaviour in left-shift
* 729321 : playbin/subtitleoverlay: Deadlock when changing subtitle track while PAUSED
* 704933 : uridecodebin: allow progressive buffering with more media types
* 720596 : discoverer: Rework the API to make " install missing plugin " feature cleaner
* 729514 : rtsp: fails to build on Windows, undefined refs to getsockname and setsockopt
* 729515 : W32: playback-test fails to build due to warnings
* 729617 : playback-test: crash when setting buffer-size property on playbin
* 729632 : rtspconnection: crashing sometimes when addinging a read source
* 730010 : gst-play: audio_sink and video_sink strings are not freed
* 730368 : Add a read source on write socket when tunnel lost.
* 730441 : dmabuf: shared the mapping with shared copies of the memory
* 729513 : W32: -base erroneously detects X11 headers from tcl/tk
==== Download ====
......@@ -176,63 +118,17 @@ subscribe to the gstreamer-devel list.
Contributors to this release
* Adrien Schwartzentruber
* Aleix Conchillo Flaque
* Aleix Conchillo Flaqué
* Alessandro Decina
* Andres Gomez
* Antoine Jacoutot
* Antonio Ospite
* Arun Raghavan
* Bastien Nocera
* Christian Fredrik Kalager Schaller
* David Svensson Fors
* Anuj Jaiswal
* Edward Hervey
* Eric Trousset
* George Kiagiadakis
* Göran Jönsson
* Haakon Sporsheim
* Hans Månsson
* Holger Kaelberer
* Jan Schmidt
* Jihyun Cho
* Johannes Dewender
* John Bassett
* Josep Torra
* Julien Isorce
* Justin Joy
* Lionel Landwerlin
* Luis de Bethencourt
* Mark Nauwelaerts
* Matej Knopp
* Mathieu Duponchelle
* MathieuDuponchelle
* Matthew Waters
* Matthieu Bouron
* Nicola Murino
* Michael Olbrich
* Nicolas Dufresne
* Ognyan Tonchev
* Olivier Crête
* Rafał Mużyło
* Ravi Kiran K N
* Reynaldo H. Verdejo Pinochet
* Sebastian Dröge
* Sebastian Rasmussen
* Sjoerd Simons
* Sreerenj Balachandran
* Stefan Sauer
* Stephan Sundermann
* Stian Selnes
* Stéphane Cerveau
* Takashi Iwai
* Thiago Santos
* Thibault Saunier
* Tim-Philipp Müller
* Todd Agulnick
* Tom Greenwood
* Vincent Penquerc'h
* William Grant
* Wim Taymans
* Wonchul Lee
* Руслан Ижбулатов
 
 
\ No newline at end of file
common @ 211fa5f2
Subproject commit 1f5d3c3163cc3399251827235355087c2affa790
Subproject commit 211fa5f2d0930dfd6891b386d42edba6d88c2a19
......@@ -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],[1.3.1.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
AC_INIT([GStreamer Base Plug-ins],[1.3.2],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
AG_GST_INIT
......@@ -56,10 +56,10 @@ dnl 1.2.5 => 205
dnl 1.10.9 (who knows) => 1009
dnl
dnl sets GST_LT_LDFLAGS
AS_LIBTOOL(GST, 301, 0, 301)
AS_LIBTOOL(GST, 302, 0, 302)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.3.1.1
GST_REQ=1.3.2
dnl *** autotools stuff ****
......
......@@ -3,7 +3,7 @@
<description>Adds multiple streams</description>
<filename>../../gst/adder/.libs/libgstadder.so</filename>
<basename>libgstadder.so</basename>
<version>1.3.1</version>
<version>1.3.2</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>1.3.1</version>
<version>1.3.2</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>1.3.1</version>
<version>1.3.2</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>1.3.1</version>
<version>1.3.2</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>1.3.1</version>
<version>1.3.2</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>1.3.1</version>
<version>1.3.2</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>1.3.1</version>
<version>1.3.2</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......