Commit 15be4ee9 authored by Jan Schmidt's avatar Jan Schmidt

configure.ac: releasing 0.10.15, "No need to argue"

Original commit message from CVS:
=== release 0.10.15 ===

2007-11-15  Jan Schmidt <jan.schmidt@sun.com>

* configure.ac:
releasing 0.10.15, "No need to argue"
parent 5424e697
=== release 0.10.15 ===
2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
* configure.ac:
releasing 0.10.15, "No need to argue"
2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
* win32/vs6/libgstfft.dsp:
This is GStreamer Base Plug-ins 0.10.14, "Light Years Ahead"
This is GStreamer Base Plug-ins 0.10.15, "No need to argue"
Please note that decodebin2 API included in this release is still
considered unstable and WILL change in future releases. At this stage, only
developers or early adopters should consider using the decodebin2 API embodied
in its signals and properties.
Changes since 0.10.14:
* RTP/RTSP/RTCP/SDP support improved
* New FFT support library libgstfft, based on Kiss FFT
* New formats supported in volume and audiotestsrc
* Fixes in audiorate and videorate
* Audio capture fixes
* Playbin and decodebin fixes
* New tagdemux base class for ID3/APE style tag readers
* Fix a nasty crash in the X sinks on shutdown
* New tags supported
* Add support for multichannel WAV files.
* Preserve channel layout information when up/down-mixing.
* Many bug-fixes and improvements
Bugs fixed since 0.10.14:
* 475395 : decodebin2 leaks request-pads
* 475451 : [decodebin2] leaks ghostpad
* 378770 : [xvimagesink] race condition in event thread?
* 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
* 430677 : [audioconvert] does not preserve channel positions when f...
* 442654 : [volume] controller bypassed by default
* 445529 : [volume] support for 24/32-bit audio/x-raw-int
* 446766 : return code for gst_base_rtp_payload_audio_handle_event()
* 451970 : Subparse requires HTML parser
* 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
* 459334 : [textoverlay] expose pango line alignment property
* 459585 : [basertpdepayload] api without namespace
* 460422 : [audiotestsrc] Add support for float and double output
* 462805 : [alsa] compilation fails with gcc 4.2
* 462979 : Add 'silent' property to GstTimeOverlay
* 463215 : [audioconvert] compile errors
* 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
* 464666 : [playbin] QT trailer hangs in preroll with decodebin2
* 464690 : Add connection-speed property to uridecodebin element
* 465015 : [playbin] Not removed probes causes deadlocks in streamin...
* 465028 : some warnings with mingw
* 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
* 468129 : [basertpaudiopayload] event handler returns the wrong value
* 468619 : New library gstfft: FFT library for integer and float typ...
* 470456 : [API] add gst_missing_*_installer_detail_new()
* 470766 : [ssaparse] line breaks in SSA subtitle parser
* 471067 : Make the SDP code useable for generating SDP descriptions
* 471194 : [rtpbuffer] RTP headers are wrong for win32
* 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
* 474384 : gstrtsp-enumtypes.c and .h needed for win32
* 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
* 475731 : rtspconnection is able to read incomplete messages
* 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
* 484989 : memleak, not unrefed caps for gstbasertppayload.c
* 489010 : Please change default channel order for WAVE_EXT-less .wa...
* 491722 : [playbin] regression: crash with external subtitles
* 492098 : [GstFFT] Broken scaling
* 492114 : Build issues on Windows/MSVC
* 492306 : compilation errors with MinGW
* 492813 : Missing symbols in libgstrtp.def
* 493986 : Build issues on Windows (missing symbols)
* 494346 : pre-release vs6 patch
* 496548 : Including malloc.h breaks macos build
* 496724 : DSW file references non-existent DSP files
* 464079 : audiotestsrc doesn't respond to conversion queries properly
* 442065 : floatcast.h includes config.h and might break other apps
* 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
* 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
* 464028 : Move connection-speed from playbin to playbasebin
API added since 0.10.14:
* GstTagDemux base class for simple tag demuxers
* GstBaseAudioSrc::provide-clock property
* gst_rtcp_ntp_to_unix()
* gst_rtcp_unix_to_ntp()
* gst_rtp_buffer_get_header_len()
* gst_rtp_buffer_get_extension_data()
* gst_rtp_buffer_compare_seqnum()
* gst_rtp_buffer_ext_timestamp()
* gst_rtcp_packet_sdes_copy_entry()
* gst_install_plugins_supported()
* gst_missing_*_installer_detail_new() convenience API
* gst_rtsp_connection_poll()
* GstTextOverlay::line-alignment property
Changes since 0.10.13:
* Audio dither and noise-shaping when reducing bit-depth
......
Release notes for GStreamer Base Plug-ins 0.10.14 "Light Years Ahead"
Release notes for GStreamer Base Plug-ins 0.10.15 "No need to argue"
......@@ -54,59 +54,89 @@ contains a set of less supported plug-ins that haven't passed the
Features of this release
* Audio dither and noise-shaping when reducing bit-depth
* RTSP and SDP helper libraries added
* Experimental buffering element "queue2" now supports pull-mode
and file-based buffering.
* Support for more 32-bit video pixel layouts
* Various fixes and improvements
* Parallel installability with 0.8.x series
* Threadsafe design and API
* RTP/RTSP/RTCP/SDP support improved
* New FFT support library libgstfft, based on Kiss FFT
* New formats supported in volume and audiotestsrc
* Fixes in audiorate and videorate
* Audio capture fixes
* Playbin and decodebin fixes
* New tagdemux base class for ID3/APE style tag readers
* Fix a nasty crash in the X sinks on shutdown
* New tags supported
* Add support for multichannel WAV files.
* Preserve channel layout information when up/down-mixing.
* Many bug-fixes and improvements
*
Bugs fixed in this release
* 380625 : [x*imagesink] add 'handle-expose' property
* 385527 : oggmux sometimes gets DELTA flag on output wrong near start
* 402076 : videoscale 4-tap method broken for downscaling
* 437169 : [xvimagesink] add property to disable Xv double-buffering
* 441264 : queue2 support to do buffering on a file
* 442553 : [v4lsrc] doesn't output segments in GST_FORMAT_TIME
* 442557 : [videorate] doesn't handle latency queries
* 442944 : Audiotestsrc can overflow on seeks
* 444523 : [queue2] Pull mode support
* 444630 : Compilation error with fsseko (from gstqueue2.c) -- unabl...
* 445505 : [queue2] It does not work in pull mode with oggdemux
* 446551 : [queue2] Buffering is not working properly if it is set t...
* 446572 : [queue2] Division by zero
* 446972 : warning when compiling gstoggdemux.c
* 449156 : Regression in CVS for decodebin2
* 450875 : Missing files in po/POTFILES.in
* 451707 : [tag] UTF-8 in ID3v1 tag not correctly decoded
* 451908 : [ffmpegcolorspace] regression: doesn't accept GST_VIDEO_C...
* 454264 : Playbin fails to " play " image url after a movie url
* 456656 : [API] Addition of audio buffer clipping function to gstaudio
* 460978 : gst_audio_buffer_clip outputs warnings
* 152864 : [PATCH] GstAlsaMixer doesn't support signals
* 360246 : [audioconvert] Optionally apply dithering
* 394061 : Add support for Subviewer subtitles
* 420326 : Base payloader class has wrong property types and ranges
* 451145 : [vorbisdec] errors out on 0-sized packets
* 459204 : [PATCH] [playbin] gst_play_base_bin_get_streaminfo_value_...
* 475395 : decodebin2 leaks request-pads
* 475451 : [decodebin2] leaks ghostpad
* 378770 : [xvimagesink] race condition in event thread?
* 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
* 430677 : [audioconvert] does not preserve channel positions when f...
* 442654 : [volume] controller bypassed by default
* 445529 : [volume] support for 24/32-bit audio/x-raw-int
* 446766 : return code for gst_base_rtp_payload_audio_handle_event()
* 451970 : Subparse requires HTML parser
* 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
* 459334 : [textoverlay] expose pango line alignment property
* 459585 : [basertpdepayload] api without namespace
* 460422 : [audiotestsrc] Add support for float and double output
* 462805 : [alsa] compilation fails with gcc 4.2
* 462979 : Add 'silent' property to GstTimeOverlay
* 463215 : [audioconvert] compile errors
* 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
* 464666 : [playbin] QT trailer hangs in preroll with decodebin2
* 464690 : Add connection-speed property to uridecodebin element
* 465015 : [playbin] Not removed probes causes deadlocks in streamin...
* 465028 : some warnings with mingw
* 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
* 468129 : [basertpaudiopayload] event handler returns the wrong value
* 468619 : New library gstfft: FFT library for integer and float typ...
* 470456 : [API] add gst_missing_*_installer_detail_new()
* 470766 : [ssaparse] line breaks in SSA subtitle parser
* 471067 : Make the SDP code useable for generating SDP descriptions
* 471194 : [rtpbuffer] RTP headers are wrong for win32
* 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
* 474384 : gstrtsp-enumtypes.c and .h needed for win32
* 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
* 475731 : rtspconnection is able to read incomplete messages
* 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
* 484989 : memleak, not unrefed caps for gstbasertppayload.c
* 489010 : Please change default channel order for WAVE_EXT-less .wa...
* 491722 : [playbin] regression: crash with external subtitles
* 492098 : [GstFFT] Broken scaling
* 492114 : Build issues on Windows/MSVC
* 492306 : compilation errors with MinGW
* 492813 : Missing symbols in libgstrtp.def
* 493986 : Build issues on Windows (missing symbols)
* 494346 : pre-release vs6 patch
* 496548 : Including malloc.h breaks macos build
* 496724 : DSW file references non-existent DSP files
* 464079 : audiotestsrc doesn't respond to conversion queries properly
* 442065 : floatcast.h includes config.h and might break other apps
* 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
* 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
* 464028 : Move connection-speed from playbin to playbasebin
API changed in this release
- API additions:
* RTSP and SDP libraries added
* gst_rtsp_base64_decode_ip
* Add buffer clipping function gst_audio_buffer_clip for raw audio buffers. Fixes #456656.
* gst_mixer_get_mixer_flags
* gst_mixer_message_parse_mute_toggled
* gst_mixer_message_parse_record_toggled
* gst_mixer_message_parse_volume_changed
* gst_mixer_message_parse_option_changed
* GstMixerMessageType
* GstMixerFlags
* GstTagDemux base class for simple tag demuxers
* GstBaseAudioSrc::provide-clock property
* gst_rtcp_ntp_to_unix()
* gst_rtcp_unix_to_ntp()
* gst_rtp_buffer_get_header_len()
* gst_rtp_buffer_get_extension_data()
* gst_rtp_buffer_compare_seqnum()
* gst_rtp_buffer_ext_timestamp()
* gst_rtcp_packet_sdes_copy_entry()
* gst_install_plugins_supported()
* gst_missing_*_installer_detail_new() convenience API
* gst_rtsp_connection_poll()
* GstTextOverlay::line-alignment property
Download
......@@ -136,19 +166,40 @@ Applications
Contributors to this release
* Andy Wingo
* Bastien Nocera
* Stefan Kost
* Alexander Shopov
* Damien Lespiau
* Dan Williams
* Daniel Díaz
* David Schleef
* Edward Hervey
* Davyd Madeley
* Funda Wang
* Haakon Sporsheim
* Ilkka Tuohela
* Jakub Bogusz
* Jan Schmidt
* Jorn Baayen
* Jason Kivlighn
* Jens Granseuer
* Johan Dahlin
* Jorge González González
* Josep Torra Valles
* Julien MOUTTE
* Laurent Glayal
* Michael Smith
* Mogens Jaeger
* Ole André Vadla Ravnås
* Olivier Crete
* Peter Kjellerstedt
* Renato Filho
* René Stadler
* Sebastian Dröge
* Sebastien Moutte
* Stefan Kost
* Thiago Sousa Santos
* Thijs Vermeir
* Thomas Vander Stichele
* Tim-Philipp Müller
* Tommi Myöhänen
* Vincent Torri
* Wim Taymans
* Yang Hong
 
\ 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, cvs and prerelease does -Werror too
dnl use a three digit version number for releases, and four for cvs/prerelease
AC_INIT(GStreamer Base Plug-ins, 0.10.14.1,
AC_INIT(GStreamer Base Plug-ins, 0.10.15,
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer,
gst-plugins-base)
......@@ -44,7 +44,7 @@ dnl - interfaces added/removed/changed -> increment CURRENT, REVISION = 0
dnl - interfaces added -> increment AGE
dnl - interfaces removed -> AGE = 0
dnl sets GST_LT_LDFLAGS
AS_LIBTOOL(GST, 10, 0, 10)
AS_LIBTOOL(GST, 11, 0, 11)
dnl FIXME: this macro doesn't actually work;
dnl the generated libtool script has no support for the listed tags.
......
......@@ -471,7 +471,7 @@
<ARG>
<NAME>GstMultiFdSink::buffers-max</NAME>
<TYPE>gint</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<RANGE>>= -1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffers max</NICK>
<BLURB>max number of buffers to queue for a client (-1 = no limit).</BLURB>
......@@ -491,7 +491,7 @@
<ARG>
<NAME>GstMultiFdSink::buffers-soft-max</NAME>
<TYPE>gint</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<RANGE>>= -1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffers soft max</NICK>
<BLURB>Recover client when going over this limit (-1 = no limit).</BLURB>
......@@ -581,7 +581,7 @@
<ARG>
<NAME>GstMultiFdSink::buffers-min</NAME>
<TYPE>gint</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<RANGE>>= -1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Buffers min</NICK>
<BLURB>min number of buffers to queue (-1 = as few as possible).</BLURB>
......@@ -611,7 +611,7 @@
<ARG>
<NAME>GstMultiFdSink::bytes-min</NAME>
<TYPE>gint</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<RANGE>>= -1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Bytes min</NICK>
<BLURB>min number of bytes to queue (-1 = as little as possible).</BLURB>
......@@ -621,7 +621,7 @@
<ARG>
<NAME>GstMultiFdSink::time-min</NAME>
<TYPE>gint64</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<RANGE>>= -1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Time min</NICK>
<BLURB>min number of time to queue (-1 = as little as possible).</BLURB>
......@@ -641,7 +641,7 @@
<ARG>
<NAME>GstMultiFdSink::units-max</NAME>
<TYPE>gint64</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<RANGE>>= -1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Units max</NICK>
<BLURB>max number of units to queue (-1 = no limit).</BLURB>
......@@ -651,7 +651,7 @@
<ARG>
<NAME>GstMultiFdSink::units-soft-max</NAME>
<TYPE>gint64</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<RANGE>>= -1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Units soft max</NICK>
<BLURB>Recover client when going over this limit (-1 = no limit).</BLURB>
......@@ -791,7 +791,7 @@
<ARG>
<NAME>GstVorbisEnc::bitrate</NAME>
<TYPE>gint</TYPE>
<RANGE>[G_MAXULONG,250001]</RANGE>
<RANGE>[-1,250001]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Target Bitrate</NICK>
<BLURB>Attempt to encode at a bitrate averaging this (in bps). This uses the bitrate management engine, and is not recommended for most users. Quality is a better alternative. (-1 == disabled).</BLURB>
......@@ -821,7 +821,7 @@
<ARG>
<NAME>GstVorbisEnc::max-bitrate</NAME>
<TYPE>gint</TYPE>
<RANGE>[G_MAXULONG,250001]</RANGE>
<RANGE>[-1,250001]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Maximum Bitrate</NICK>
<BLURB>Specify a maximum bitrate (in bps). Useful for streaming applications. (-1 == disabled).</BLURB>
......@@ -831,7 +831,7 @@
<ARG>
<NAME>GstVorbisEnc::min-bitrate</NAME>
<TYPE>gint</TYPE>
<RANGE>[G_MAXULONG,250001]</RANGE>
<RANGE>[-1,250001]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Minimum Bitrate</NICK>
<BLURB>Specify a minimum bitrate (in bps). Useful for encoding for a fixed-size channel. (-1 == disabled).</BLURB>
......@@ -1428,6 +1428,26 @@
<DEFAULT>baseline</DEFAULT>
</ARG>
<ARG>
<NAME>GstTextOverlay::line-alignment</NAME>
<TYPE>GstTextOverlayLineAlign</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>line alignment</NICK>
<BLURB>Alignment of text lines relative to each other.</BLURB>
<DEFAULT>center</DEFAULT>
</ARG>
<ARG>
<NAME>GstTextOverlay::silent</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>silent</NICK>
<BLURB>Whether to render the text string.</BLURB>
<DEFAULT>FALSE</DEFAULT>
</ARG>
<ARG>
<NAME>CDParanoia::abort-on-skip</NAME>
<TYPE>gboolean</TYPE>
......@@ -1621,7 +1641,7 @@
<ARG>
<NAME>GstCdParanoiaSrc::read-speed</NAME>
<TYPE>gint</TYPE>
<RANGE>>= G_MAXULONG</RANGE>
<RANGE>>= -1</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Read speed</NICK>
<BLURB>Read from device at specified speed.</BLURB>
......@@ -1631,7 +1651,7 @@
<ARG>
<NAME>GstCdParanoiaSrc::search-overlap</NAME>
<TYPE>gint</TYPE>
<RANGE>[G_MAXULONG,75]</RANGE>
<RANGE>[-1,75]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Search overlap</NICK>
<BLURB>Force minimum overlap search during verification to n sectors.</BLURB>
......@@ -1698,6 +1718,16 @@
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstDecodeBin2::subtitle-encoding</NAME>
<TYPE>gchararray</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>subtitle encoding</NICK>
<BLURB>Encoding to assume if input subtitles are not in UTF-8 encoding. If not set, the GST_SUBTITLE_ENCODING environment variable will be checked for an encoding to use. If that is not set either, ISO-8859-15 will be assumed.</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
<ARG>
<NAME>GstURIDecodeBin::uri</NAME>
<TYPE>gchararray</TYPE>
......@@ -1718,6 +1748,26 @@
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstURIDecodeBin::caps</NAME>
<TYPE>GstCaps</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Caps</NICK>
<BLURB>The caps on which to stop decoding. (NULL = default).</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG>
<NAME>GstURIDecodeBin::subtitle-encoding</NAME>
<TYPE>gchararray</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>subtitle encoding</NICK>
<BLURB>Encoding to assume if input subtitles are not in UTF-8 encoding. If not set, the GST_SUBTITLE_ENCODING environment variable will be checked for an encoding to use. If that is not set either, ISO-8859-15 will be assumed.</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
<ARG>
<NAME>GstQueue2::current-level-buffers</NAME>
<TYPE>guint</TYPE>
......
......@@ -151,7 +151,8 @@ gint arg1
<RETURNS>gboolean</RETURNS>
<FLAGS>l</FLAGS>
GstDecodeBin2 *gstdecodebin2
GstCaps *arg1
GstPad *arg1
GstCaps *arg2
</SIGNAL>
<SIGNAL>
......@@ -189,3 +190,59 @@ GstPad *arg1
GstCaps *arg2
</SIGNAL>
<SIGNAL>
<NAME>GstDecodeBin2::autoplug-factories</NAME>
<RETURNS>GValueArray*</RETURNS>
<FLAGS>l</FLAGS>
GstDecodeBin2 *gstdecodebin2
GstPad *arg1
GstCaps *arg2
</SIGNAL>
<SIGNAL>
<NAME>GstDecodeBin2::autoplug-select</NAME>
<RETURNS>gint</RETURNS>
<FLAGS>l</FLAGS>
GstDecodeBin2 *gstdecodebin2
GstPad *arg1
GstCaps *arg2
GValueArray *arg3
</SIGNAL>
<SIGNAL>
<NAME>GstURIDecodeBin::autoplug-continue</NAME>
<RETURNS>gboolean</RETURNS>
<FLAGS>l</FLAGS>
GstURIDecodeBin *gsturidecodebin
GstPad *arg1
GstCaps *arg2
</SIGNAL>
<SIGNAL>
<NAME>GstURIDecodeBin::autoplug-factories</NAME>
<RETURNS>GValueArray*</RETURNS>
<FLAGS>l</FLAGS>
GstURIDecodeBin *gsturidecodebin
GstPad *arg1
GstCaps *arg2
</SIGNAL>
<SIGNAL>
<NAME>GstURIDecodeBin::autoplug-select</NAME>
<RETURNS>gint</RETURNS>
<FLAGS>l</FLAGS>
GstURIDecodeBin *gsturidecodebin
GstPad *arg1
GstCaps *arg2
GValueArray *arg3
</SIGNAL>
<SIGNAL>
<NAME>GstURIDecodeBin::unknown-type</NAME>
<RETURNS>void</RETURNS>
<FLAGS>l</FLAGS>
GstURIDecodeBin *gsturidecodebin
GstPad *arg1
GstCaps *arg2
</SIGNAL>
......@@ -3,7 +3,7 @@
<description>Adds multiple streams</description>
<filename>../../gst/adder/.libs/libgstadder.so</filename>
<basename>libgstadder.so</basename>
<version>0.10.14</version>
<version>0.10.15</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>0.10.14</version>
<version>0.10.15</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......@@ -30,7 +30,7 @@
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
<details>audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
<details>audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
</caps>
</pads>
</element>
......@@ -45,7 +45,7 @@
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
<details>audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)32, depth=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)32, depth=(int)24, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)24, depth=(int)24, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, endianness=(int){ 1234, 4321 }, signed=(boolean){ true, false }, width=(int)16, depth=(int)16, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]; audio/x-raw-int, signed=(boolean){ true, false }, width=(int)8, depth=(int)8, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
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......
......@@ -3,7 +3,7 @@
<description>Convert audio to different formats</description>
<filename>../../gst/audioconvert/.libs/libgstaudioconvert.so</filename>
<basename>libgstaudioconvert.so</basename>
<version>0.10.14</version>
<version>0.10.15</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>0.10.14</version>
<version>0.10.15</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>0.10.14</version>
<version>0.10.15</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>0.10.14</version>
<version>0.10.15</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......@@ -20,7 +20,7 @@
<name>src</name>
<direction>source</direction>
<presence>always</presence>
<details>audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)[ 1, 2147483647 ], channels=(int)1</details>
<details>audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)[ 1, 2147483647 ], channels=(int)1; audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)[ 1, 2147483647 ], channels=(int)1; audio/x-raw-float, endianness=(int)1234, width=(int){ 32, 64 }, rate=(int)[ 1, 2147483647 ], channels=(int)1</details>
</caps>
</pads>
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......
......@@ -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>0.10.14</version>
<version>0.10.15</version>
<license>GPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>decoder bin</description>
<filename>../../gst/playback/.libs/libgstdecodebin.so</filename>
<basename>libgstdecodebin.so</basename>
<version>0.10.14</version>
<version>0.10.15</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>decoder bin newer version</description>
<filename>../../gst/playback/.libs/libgstdecodebin2.so</filename>
<basename>libgstdecodebin2.so</basename>
<version>0.10.14</version>
<version>0.10.15</version>
<license>LGPL</license>
<source>gst-plugins-base</source>
<package>GStreamer Base Plug-ins source release</package>
......
......@@ -3,7 +3,7 @@
<description>colorspace conversion copied from FFMpeg 0.4.9-pre1</description>
<filename>../../gst/ffmpegcolorspace/.libs/libgstffmpegcolorspace.so</filename>