mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 02:01:12 +00:00
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"
This commit is contained in:
parent
5424e697fb
commit
15be4ee905
35 changed files with 377 additions and 107 deletions
|
@ -1,3 +1,10 @@
|
|||
=== 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:
|
||||
|
|
85
NEWS
85
NEWS
|
@ -1,10 +1,93 @@
|
|||
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
|
||||
|
|
153
RELEASE
153
RELEASE
|
@ -1,5 +1,5 @@
|
|||
|
||||
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
|
||||
|
|
@ -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>
|
||||
</caps>
|
||||
</pads>
|
||||
</element>
|
||||
|
|
|
@ -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>
|
||||
</element>
|
||||
|
|
|
@ -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>
|
||||
<basename>libgstffmpegcolorspace.so</basename>
|
||||
<version>0.10.14</version>
|
||||
<version>0.10.15</version>
|
||||
<license>LGPL</license>
|
||||
<source>gst-plugins-base</source>
|
||||
<package>FFMpeg</package>
|
||||
|
|
|
@ -3,7 +3,7 @@
|
|||
<description>Payload/depayload GDP packets</description>
|
||||
<filename>../../gst/gdp/.libs/libgstgdp.so</filename>
|
||||
<basename>libgstgdp.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>elements to read from and write to Gnome-VFS uri's</description>
|
||||
<filename>../../ext/gnomevfs/.libs/libgstgnomevfs.so</filename>
|
||||
<basename>libgstgnomevfs.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>libvisual visualization plugins</description>
|
||||
<filename>../../ext/libvisual/.libs/libgstlibvisual.so</filename>
|
||||
<basename>libgstlibvisual.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>
|
||||
|
@ -114,6 +114,27 @@
|
|||
</caps>
|
||||
</pads>
|
||||
</element>
|
||||
<element>
|
||||
<name>libvisual_lv_analyzer</name>
|
||||
<longname>libvisual libvisual analyzer plugin v.1.0</longname>
|
||||
<class>Visualization</class>
|
||||
<description>Libvisual analyzer plugin</description>
|
||||
<author>Benjamin Otte <otte@gnome.org></author>
|
||||
<pads>
|
||||
<caps>
|
||||
<name>src</name>
|
||||
<direction>source</direction>
|
||||
<presence>always</presence>
|
||||
<details>video/x-raw-rgb, bpp=(int)32, depth=(int)24, endianness=(int)4321, red_mask=(int)65280, green_mask=(int)16711680, blue_mask=(int)-16777216, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw-rgb, bpp=(int)24, depth=(int)24, endianness=(int)4321, red_mask=(int)255, green_mask=(int)65280, blue_mask=(int)16711680, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw-rgb, bpp=(int)16, depth=(int)16, endianness=(int)1234, red_mask=(int)63488, green_mask=(int)2016, blue_mask=(int)31, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
|
||||
</caps>
|
||||
<caps>
|
||||
<name>sink</name>
|
||||
<direction>sink</direction>
|
||||
<presence>always</presence>
|
||||
<details>audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int)1234, signed=(boolean)true, channels=(int){ 1, 2 }, rate=(int){ 8000, 11250, 22500, 32000, 44100, 48000, 96000 }</details>
|
||||
</caps>
|
||||
</pads>
|
||||
</element>
|
||||
<element>
|
||||
<name>libvisual_lv_scope</name>
|
||||
<longname>libvisual libvisual scope plugin v.0.1</longname>
|
||||
|
|
|
@ -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>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>Pango-based text rendering and overlay</description>
|
||||
<filename>../../ext/pango/.libs/libgstpango.so</filename>
|
||||
<basename>libgstpango.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>player bin</description>
|
||||
<filename>../../gst/playback/.libs/libgstplaybin.so</filename>
|
||||
<basename>libgstplaybin.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>Subtitle parsing</description>
|
||||
<filename>../../gst/subparse/.libs/libgstsubparse.so</filename>
|
||||
<basename>libgstsubparse.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>transfer data over the network via TCP</description>
|
||||
<filename>../../gst/tcp/.libs/libgsttcp.so</filename>
|
||||
<basename>libgsttcp.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>Theora plugin library</description>
|
||||
<filename>../../ext/theora/.libs/libgsttheora.so</filename>
|
||||
<basename>libgsttheora.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>default typefind functions</description>
|
||||
<filename>../../gst/typefind/.libs/libgsttypefindfunctions.so</filename>
|
||||
<basename>libgsttypefindfunctions.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>elements for Video 4 Linux</description>
|
||||
<filename>../../sys/v4l/.libs/libgstvideo4linux.so</filename>
|
||||
<basename>libgstvideo4linux.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 video frames</description>
|
||||
<filename>../../gst/videorate/.libs/libgstvideorate.so</filename>
|
||||
<basename>libgstvideorate.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>Resizes video</description>
|
||||
<filename>../../gst/videoscale/.libs/libgstvideoscale.so</filename>
|
||||
<basename>libgstvideoscale.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 a test video stream</description>
|
||||
<filename>../../gst/videotestsrc/.libs/libgstvideotestsrc.so</filename>
|
||||
<basename>libgstvideotestsrc.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>plugin for controlling audio volume</description>
|
||||
<filename>../../gst/volume/.libs/libgstvolume.so</filename>
|
||||
<basename>libgstvolume.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>
|
||||
|
@ -20,13 +20,13 @@
|
|||
<name>src</name>
|
||||
<direction>source</direction>
|
||||
<presence>always</presence>
|
||||
<details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int){ 32, 64 }; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true</details>
|
||||
<details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int){ 32, 64 }; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)8, depth=(int)8, signed=(boolean)true; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)24, depth=(int)24, signed=(boolean)true; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)32, depth=(int)32, signed=(boolean)true</details>
|
||||
</caps>
|
||||
<caps>
|
||||
<name>sink</name>
|
||||
<direction>sink</direction>
|
||||
<presence>always</presence>
|
||||
<details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int){ 32, 64 }; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true</details>
|
||||
<details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int){ 32, 64 }; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)8, depth=(int)8, signed=(boolean)true; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)24, depth=(int)24, signed=(boolean)true; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)32, depth=(int)32, signed=(boolean)true</details>
|
||||
</caps>
|
||||
</pads>
|
||||
</element>
|
||||
|
|
|
@ -3,7 +3,7 @@
|
|||
<description>Vorbis plugin library</description>
|
||||
<filename>../../ext/vorbis/.libs/libgstvorbis.so</filename>
|
||||
<basename>libgstvorbis.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>X11 video output element based on standard Xlib calls</description>
|
||||
<filename>../../sys/ximage/.libs/libgstximagesink.so</filename>
|
||||
<basename>libgstximagesink.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>XFree86 video output plugin using Xv extension</description>
|
||||
<filename>../../sys/xvimage/.libs/libgstxvimagesink.so</filename>
|
||||
<basename>libgstxvimagesink.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>
|
||||
|
|
|
@ -169,10 +169,10 @@
|
|||
#undef HAVE_SYS_SOCKET_H
|
||||
|
||||
/* Define to 1 if you have the <sys/stat.h> header file. */
|
||||
#undef HAVE_SYS_STAT_H 1
|
||||
#undef HAVE_SYS_STAT_H
|
||||
|
||||
/* Define to 1 if you have the <sys/types.h> header file. */
|
||||
#undef HAVE_SYS_TYPES_H 1
|
||||
#undef HAVE_SYS_TYPES_H
|
||||
|
||||
/* support for features: theoradec theoraenc */
|
||||
#undef HAVE_THEORA
|
||||
|
@ -208,16 +208,16 @@
|
|||
#define PACKAGE_BUGREPORT "http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer"
|
||||
|
||||
/* Define to the full name of this package. */
|
||||
#undef PACKAGE_NAME "GStreamer Base Plug-ins"
|
||||
#define PACKAGE_NAME "GStreamer Base Plug-ins"
|
||||
|
||||
/* Define to the full name and version of this package. */
|
||||
#undef PACKAGE_STRING "GStreamer Base Plug-ins 0.10.14"
|
||||
#define PACKAGE_STRING "GStreamer Base Plug-ins 0.10.15"
|
||||
|
||||
/* Define to the one symbol short name of this package. */
|
||||
#undef PACKAGE_TARNAME "gst-plugins-base"
|
||||
#define PACKAGE_TARNAME "gst-plugins-base"
|
||||
|
||||
/* Define to the version of this package. */
|
||||
#undef PACKAGE_VERSION "0.10.14"
|
||||
#define PACKAGE_VERSION "0.10.15"
|
||||
|
||||
/* directory where plugins are located */
|
||||
#undef PLUGINDIR
|
||||
|
@ -241,7 +241,7 @@
|
|||
#undef STDC_HEADERS
|
||||
|
||||
/* Version number of package */
|
||||
#define VERSION "0.10.14"
|
||||
#define VERSION "0.10.15"
|
||||
|
||||
/* Define to 1 if your processor stores words with the most significant byte
|
||||
first (like Motorola and SPARC, unlike Intel and VAX). */
|
||||
|
@ -256,5 +256,6 @@
|
|||
#undef inline
|
||||
#endif
|
||||
|
||||
/* FIXME: this should probably be hard-coded to some win32 system path */
|
||||
#define GST_INSTALL_PLUGINS_HELPER "/home/jan/.install/libexec/gst-install-plugins-helper"
|
||||
|
||||
|
|
Loading…
Reference in a new issue