Commit 68f5350c authored by Sebastian Dröge's avatar Sebastian Dröge

Release 1.3.1

parent 876e28b9
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This is GStreamer Base Plugins 1.2.0
This is GStreamer Base Plugins 1.3.1
Changes since 1.0:
Changes since 1.2:
New API:
• GstContext negotiation / sharing / announcing for sharing a
generic context between elements, e.g. a display handle
• GL texture upload conversion meta for allowing different
buffer types to be converted to an OpenGL texture
• GstCapsFeatures as extension to GstCaps for allowing the
negotiation of specific memory or meta requirements between
elements
• GstMemory flags for contiguous and non-mappable memory
• The stream-start event has optional flags now, e.g. for signalling
sparse streams
• The stream-start even has an optional group-id field now to signal
all streams that should be played together
• Allocators library in gst-plugins-base, currently only with generic
dmabuf memory support
• insertbin library for easier handling of dynamically linked
pipelines (in -bad for now)
• EGL helper library (in -bad for now)
• MPEG-TS data structure library (in -bad for now)
• New GstVideoRegionOfInterestMeta to describe a region of interest on
video frames.
• GstVideoDecoder/Encoder has new ::flush() vfunc to replace the
ill-defined ::reset() vfunc.
• The URI query allows to query the redirected URI now.
• GstMessageType has GST_MESSAGE_EXTENDED added. All types before
that can be used together as a flags type as before, but from
that message onwards the types are just counted incrementally.
This was necessary to be able to add more message types.
In 2.0 GstMessageType will just become an enum and not a flags
type anymore.
• GstDeviceMonitor for device probing, e.g. to list all available
audio or video capture devices. This is the replacement for
GstPropertyProbe from 0.10.
• Events accumulate the running-time offset now when travelling
through pads, as set by the gst_pad_set_offset() function. This
allows to compensate for this in the QOS event for example.
• GstBuffer has a new flag "tag-memory" that is set automatically
when memory is added or removed to a buffer. This allows buffer
pools to detect if they can recycle a buffer or need to reset
it first.
• GstToc has new API to mark GstTocEntries as loops.
• A not-authorized resource error has been defined to notify
applications that accessing the resource has failed because
of missing authorization and to distinguish this case from others.
This change is actually already in 1.2.4.
• GstPad has a new flag "accept-intersect", that will let the default
ACCEPT_CAPS query handler do an intersection instead of subset check.
This is interesting for parser elements that can handle incomplete
caps.
• GstCollectPads has support for flushing and a default handler for
SEEK events now.
• GstSegment has new API to offset the running time by a specific
value and this is used in GstPad to allow positive and negative
offsets in gst_pad_set_offset() in all situations.
• Support for h265/HEVC and VP8 has been added to the codec utils and codec
parsers library, and was integrated into various elements.
• API for adjusting the TLS validation of RTSP connection has been added.
• The RTSP and SDP library has MIKEY (RFC 3830) support now, and
there is API to distinguish between the different RTSP profiles.
• API to access RTP time information and statistics.
• Support for auxiliary streams was added to rtpbin.
• Support for tiled, raw video formats has been added.
• GstVideoDecoder and GstAudioDecoder have API to help aggregating tag
events and merge custom tags into them consistently.
• playbin/playsink has support for application provided audio and video
filters.
• 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
a video sink. Supported platforms currently are Android, Cocoa (OS X),
DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11,
Wayland and EGL platforms.
This replaces eglglessink and also is supposed to replace osxvideosink.
Major changes:
• New tool: gst-play-1.0 in gst-plugins-base for basic playback
testing on the command line.
• New plugins:
∘ mssdemux for Microsoft Smooth Streaming
∘ dashdemux for DASH adaptive streaming protocol
∘ bluez for interaction with Bluetooth devices
∘ openjpeg for JPEG2000 decoding and encoding
∘ daala for experimental Daala decoding and encoding
∘ vpx plugin has experimental VP9 decoding and encoding support
∘ webp plugin for WebP decoding (encoding to be added later)
∘ Various others: yadif, srtp, sbc, fluidsynth, midiparse,
mfc, ivtv, accuraterip and audiofxbad
• Moved plugins:
∘ dtmf, vp8rtp, scaletempo and rtpmux plugins are in
gst-plugins-good now
• Video:
∘ Fix handling of interlaced video in converters such as videoscale
and videoconvert (e.g. scale both fields independently)
∘ videoconvert will try harder to minimise quality losses when
conversion is necessary
∘ The experimental GstSurfaceConverter, GstSurfaceMeta and
GstVideoContext APIs from the (confusingly-named)
libgstbasevideo-1.0 library in gst-plugins-bad have now been
removed and been replaced by new APIs in GStreamer Core and
gst-plugins-base (see above). Since that was all that was left in
this library, the entire experimental libgstbasevideo-1.0 library
has been removed from gst-plugins-bad
∘ Chroma subsampling and chroma siting conversion is better handled
in videoconvert and the support for interlaced video was improved.
∘ New pinwheel and spoke patterns in videotestsrc
∘ videomixer can now accept different video formats on its sinkpads
and converts to a common format during mixing
• Audio:
∘ audioconvert will try harder to minimise quality losses when
conversion is necessary
∘ adder now allows muting/unmuting of its input streams, and also
per-input stream volume
∘ pulseaudio elements can switch between devices during playback now
∘ aacparse can convert between ADTS←→RAW
• Platform specific changes:
∘ Caps, events, etc. are now printed in the GStreamer debug logs
with their content instead of just the pointer address even on
non-glibc platforms (e.g. Windows, OSX, Android).
∘ Network elements (UDP/TCP) now work better with platforms,
where IPv6 sockets can't handle IPv4 (e.g. Windows)
∘ Linux/BSD: v4l2 had many improvements and cleanups
Major changes:
• New plugins and elements:
∘ v4l2videodec element for accessing hardware codecs on
platforms that make them accessible via V4L2, e.g.
Samsung Exynos. This comes together with major refactoring
of the existing V4L2 elements and the corresponding
infrastructure.
The v4l2videodec element replaces the mfcdec element.
∘ rtpstreampay and rtpstreamdepay elements for transmitting
RTP packets over a stream API (e.g. TCP) according to
RFC 4571.
∘ rtprtx elements for standard compliant implementation of
retransmissions, integrated into the rtpmanager plugin.
∘ audiomixer element that mixes multiple audio streams together
into a single one while keeping synchronization. This is
planned to become the replacement of the adder element.
∘ OpenNI2 plugin for 3D cameras like the Kinect camera.
∘ OpenEXR plugin for decoding high-dynamic-range EXR images.
∘ curlsshsink and curlsftpsink to write files via SSH/SFTP.
∘ videosignal, ivfparse and sndfile plugins ported from 0.10.
∘ avfvideosrc, vtdec and other elements were ported from 0.10 and
are available on OS X and iOS now.
• Other changes:
∘ gst-libav now uses libav 9
∘ Static linking of plugins is supported now (also in 1.0.7)
∘ rtspsrc: add support for NetClientClock: when the server suggests a
GstNetTimeProvider in the SDP, set up a GstNetClientClock that
slaves to the remote clock and suggest this clock in provide_clock.
Simplifies synchronized playback of a resource from an RTSP server.
gst-rtsp-server now supports adding this to the SDP and can provide
a network clock
∘ RTP retransmission / NACK support and big RTP jitterbuffer improvements
∘ SRTP and DTLS support
∘ Changes to many elements and core to use the correct sticky event
order and also not lose any important sticky events during flushing
∘ >1000 fixed bug reports, and many other bug fixes and other
improvements everywhere that had no bug report
∘ gst-libav now uses libav 10, 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.
∘ Various bugfixes and improvements to buffering in queue2 and
multiqueue elements.
∘ dvbsrc supports more delivery mechanisms and other features
now, including DVB S2 and T2 support.
∘ The MPEGTS library has support for many more descriptors.
∘ Major improvements to tsdemux, especially time related.
∘ souphttpsrc now has support for keep-alive connections,
compression, configurable number of retries and configuration
for SSL certificate validation.
∘ hlsdemux has undergone major refactoring and works more
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.
∘ videoflip can automatically flip based on the orientation tag.
∘ openjpeg supports the OpenJPEG2 API.
∘ gst-rtsp-server supports SRTP and MIKEY now.
∘ Lots of fixes for coverity warnings all over the place.
∘ 400+ fixed bug reports, and many other bug fixes and other
improvements everywhere that had no bug report.
Things to look out for:
• Single header includes for all libraries, e.g. #include
<gst/video/video.h> - this was needed for some bindings.
• Stricter (correct) caps subset checking in some cases where this was
not correct before. Caps will now always fail to be a compatible
subset of another set of caps if the subset caps are missing some
fields that the superset caps have. This might lead to not-negotiated
errors if caps are incomplete now. However, it also prevents possible
data corruption caused by piping data formatted in an
incompatible/unexpected way into some elements. Check your h264 caps
for stream-format and alignment fields and AAC caps for the
stream-format field. This change will also be included in the next
stable 1.0.8 release.
• Stricter checking for missing events and correct sticky event order
(stream-start, caps, segment) in some places; this is not enabled in
stable releases by default, but you may get warnings when using git
builds, development releases or when compiling with
-UG_DISABLE_ASSERT in CFLAGS
• x264enc now outputs data in byte-stream by default if downstream has
ANY caps (e.g. appsink without caps set, filesink, udpsink,
tcpserversink etc.)
• The MPEG TS demuxer posts messages contain the PMT, PAT, etc. in a
different format now. This new format uses the data structures from
the new MPEGTS library
• The GstContext API has changed between 1.1.4 and 1.1.90
• The eglglessink element was removed and replaced by the glimagesink
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 1.2.0
The GStreamer team is pleased to announce the first 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
will lead to the stable 1.4 release series in the next weeks, and newly added
API can still change until that point.
The GStreamer team is proud to announce a new feature release
in the 1.x stable series of the
core of the GStreamer streaming media framework.
The 1.x series is a stable series targeted at end users.
It is not API or ABI compatible with the stable 0.10.x series.
It is, however, parallel installable with the 0.10.x series and
will not affect an existing 0.10.x installation.
Binaries for Android, iOS, Mac OS X and Windows will be provided separately
during the unstable 1.3 release series.
The versioning scheme that is used in general is that 1.x.y is API and
ABI backwards compatible with previous 1.x.y releases. If x is an even
number it is a stable release series and all releases in this series
will only contain important bugfixes, e.g. the 1.0 series with 1.0.7. If
x is odd it is a development release series that will lead to the next
stable release series 1.x+1 and contains new features and bigger
changes. During the development release series, new API can still
change.
......@@ -54,16 +66,80 @@ 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
* 708667 : rtspconnection: leaks file descriptors/child sources
* 708372 : dmabuf: sys/mman.h: No such file or directory
* 708590 : adder: Should send its segment before checking for eos
* 708606 : video-frame: offsets are not copied from metadata
* 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
==== Download ====
......@@ -100,10 +176,63 @@ 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
* 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
* 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
 
\ No newline at end of file
* Wonchul Lee
* Руслан Ижбулатов
 
......@@ -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],[1.3.0.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-base])
AC_INIT([GStreamer Base Plug-ins],[1.3.1],[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, 300, 0, 300)
AS_LIBTOOL(GST, 301, 0, 301)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.3.0.1
GST_REQ=1.3.1
dnl *** autotools stuff ****
......
......@@ -338,6 +338,26 @@
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstPlaySink::audio-filter</NAME>
<TYPE>GstElement*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Audio filter</NICK>
<BLURB>the audio filter(s) to apply, if possible.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstPlaySink::video-filter</NAME>
<TYPE>GstElement*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Video filter</NICK>
<BLURB>the video filter(s) to apply, if possible.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstPlayBin::audio-sink</NAME>
<TYPE>GstElement*</TYPE>
......@@ -638,6 +658,26 @@
<DEFAULT>1</DEFAULT>
</ARG>
<ARG>
<NAME>GstPlayBin::audio-filter</NAME>
<TYPE>GstElement*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Audio filter</NICK>
<BLURB>the audio filter(s) to apply, if possible.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstPlayBin::video-filter</NAME>
<TYPE>GstElement*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Video filter</NICK>
<BLURB>the video filter(s) to apply, if possible.</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstDecodeBin::caps</NAME>
<TYPE>GstCaps*</TYPE>
......
......@@ -108,6 +108,7 @@ GObject
GstPadTemplate
GstPlugin
GstPluginFeature
GstDeviceMonitorFactory
GstElementFactory
GstTypeFindFactory
GstRegistry
......
......@@ -3,10 +3,10 @@
<description>Adds multiple streams</description>
<filename>../../gst/adder/.libs/libgstadder.so</filename>
<basename>libgstadder.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>ALSA plugin library</description>
<filename>../../ext/alsa/.libs/libgstalsa.so</filename>
<basename>libgstalsa.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Elements used to communicate with applications</description>
<filename>../../gst/app/.libs/libgstapp.so</filename>
<basename>libgstapp.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Convert audio to different formats</description>
<filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
<basename>libgstaudioconvert.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Adjusts audio frames</description>
<filename>../../gst/audiorate/.libs/libgstaudiorate.so</filename>
<basename>libgstaudiorate.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Resamples audio</description>
<filename>../../gst/audioresample/.libs/libgstaudioresample.so</filename>
<basename>libgstaudioresample.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Creates audio test signals of given frequency and volume</description>
<filename>../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so</filename>
<basename>libgstaudiotestsrc.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Read audio from CD in paranoid mode</description>
<filename>../../ext/cdparanoia/.libs/libgstcdparanoia.so</filename>
<basename>libgstcdparanoia.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>various encoding-related elements</description>
<filename>../../gst/encoding/.libs/libgstencodebin.so</filename>
<basename>libgstencodebin.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>GIO elements</description>
<filename>../../gst/gio/.libs/libgstgio.so</filename>
<basename>libgstgio.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,7 +3,7 @@
<description>Vorbis Tremor decoder</description>
<filename>../../ext/vorbis/.libs/libgstivorbisdec.so</filename>
<basename>libgstivorbisdec.so</basename>
<version>1.2.0</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,10 +3,10 @@
<description>libvisual visualization plugins</description>
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
<basename>libgstlibvisual.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>ogg stream manipulation (info about ogg: http://xiph.org)</description>
<filename>../../ext/ogg/.libs/libgstogg.so</filename>
<basename>libgstogg.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Pango-based text rendering and overlay</description>
<filename>../../ext/pango/.libs/libgstpango.so</filename>
<basename>libgstpango.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>various playback elements</description>
<filename>../../gst/playback/.libs/libgstplayback.so</filename>
<basename>libgstplayback.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins git</package>
<package>GStreamer Base Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
......
......@@ -3,10 +3,10 @@
<description>Subtitle parsing</description>
<filename>../../gst/subparse/.libs/libgstsubparse.so</filename>
<basename>libgstsubparse.so</basename>
<version>1.3.0.1</version>
<version>1.3.1</version>
<license>LGPL</license>