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update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
@comment $NetBSD: PLIST,v 1.22 2010/12/03 18:15:44 drochner Exp $
bin/gst-discoverer-0.10
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
bin/gst-visualise-0.10
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
include/gstreamer-0.10/gst/app/gstappbuffer.h
include/gstreamer-0.10/gst/app/gstappsink.h
include/gstreamer-0.10/gst/app/gstappsrc.h
include/gstreamer-0.10/gst/audio/audio-enumtypes.h
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
include/gstreamer-0.10/gst/audio/audio.h
include/gstreamer-0.10/gst/audio/gstaudioclock.h
include/gstreamer-0.10/gst/audio/gstaudiofilter.h
include/gstreamer-0.10/gst/audio/gstaudiosink.h
include/gstreamer-0.10/gst/audio/gstaudiosrc.h
include/gstreamer-0.10/gst/audio/gstbaseaudiosink.h
include/gstreamer-0.10/gst/audio/gstbaseaudiosrc.h
include/gstreamer-0.10/gst/audio/gstringbuffer.h
Update to 0.10.2, ok jmmv@: Changes since 0.10.1: * ported gnomevfssink, cdparanoia * New library and base class: GstCddaBaseSrc * ported mixerutils.h * added 'sine-tab' waveform to audiotestsrc * added float audio to audiorate Bugs fixed since 0.10.1: * 324216 : [cdparanoia] missing patches from 0.8 * 324696 : [videotestsrc] does not start counting the time from zero... * 324900 : Problem compiling gst-plugins-base with Forte * 325984 : [playbin] cannot handle sources that produce raw audio/video * 325990 : patch videotestsrc for using glib types * 326601 : GstRingBuffer crashes with alaw/mulaw caps * 327114 : [theoradec] should post tags on the bus * 327216 : vorbisdec segfaults on certain queries
2006-01-25 20:52:47 +01:00
include/gstreamer-0.10/gst/audio/mixerutils.h
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
include/gstreamer-0.10/gst/audio/multichannel.h
Update to 0.10.2, ok jmmv@: Changes since 0.10.1: * ported gnomevfssink, cdparanoia * New library and base class: GstCddaBaseSrc * ported mixerutils.h * added 'sine-tab' waveform to audiotestsrc * added float audio to audiorate Bugs fixed since 0.10.1: * 324216 : [cdparanoia] missing patches from 0.8 * 324696 : [videotestsrc] does not start counting the time from zero... * 324900 : Problem compiling gst-plugins-base with Forte * 325984 : [playbin] cannot handle sources that produce raw audio/video * 325990 : patch videotestsrc for using glib types * 326601 : GstRingBuffer crashes with alaw/mulaw caps * 327114 : [theoradec] should post tags on the bus * 327216 : vorbisdec segfaults on certain queries
2006-01-25 20:52:47 +01:00
include/gstreamer-0.10/gst/cdda/gstcddabasesrc.h
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
include/gstreamer-0.10/gst/fft/gstfft.h
include/gstreamer-0.10/gst/fft/gstfftf32.h
include/gstreamer-0.10/gst/fft/gstfftf64.h
include/gstreamer-0.10/gst/fft/gstffts16.h
include/gstreamer-0.10/gst/fft/gstffts32.h
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
include/gstreamer-0.10/gst/floatcast/floatcast.h
include/gstreamer-0.10/gst/interfaces/colorbalance.h
include/gstreamer-0.10/gst/interfaces/colorbalancechannel.h
include/gstreamer-0.10/gst/interfaces/interfaces-enumtypes.h
include/gstreamer-0.10/gst/interfaces/mixer.h
include/gstreamer-0.10/gst/interfaces/mixeroptions.h
include/gstreamer-0.10/gst/interfaces/mixertrack.h
include/gstreamer-0.10/gst/interfaces/navigation.h
include/gstreamer-0.10/gst/interfaces/propertyprobe.h
update to 0.10.25 changes: Add per-stream volume controls Theora 1.0 and Y444 and Y42B format support Improve audio capture timing GObject introspection support Improve audio output startup RTSP improvements Use pango-cairo instead of pangoft2 Allow cdda://(device#)?track URI scheme in cddabasesrc Support interlaced content in videoscale and ffmpegcolorspacee Many other bug fixes and improvements
2009-10-21 18:45:31 +02:00
include/gstreamer-0.10/gst/interfaces/streamvolume.h
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
include/gstreamer-0.10/gst/interfaces/tuner.h
include/gstreamer-0.10/gst/interfaces/tunerchannel.h
include/gstreamer-0.10/gst/interfaces/tunernorm.h
update to 0.10.11 changes: This release requires GStreamer core 0.10.11 or higher Improved buffering during network playback Support for reverse playback of Ogg/Vorbis/Theora Reverse playback support in the audio sinks The Ogg muxer now produces correct streams in all cases Support libvisual 0.4 Improvements in RTP support Support retagging of Vorbis files Introduction of the decodebin2 element Many other bug fixes and improvements
2006-12-11 13:46:08 +01:00
include/gstreamer-0.10/gst/interfaces/videoorientation.h
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
include/gstreamer-0.10/gst/interfaces/xoverlay.h
include/gstreamer-0.10/gst/netbuffer/gstnetbuffer.h
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
include/gstreamer-0.10/gst/pbutils/codec-utils.h
update to 0.10.12 changes: -New API for on-demand plugin installation -Xv thread-safety and configuration enhancements -decodebin2 improvements -Support more raw audio format conversions -Improvements in Ogg support -AudioFilter base class ported to 0.10 -Fixes for subtitles -Latency/live-playback support for Alsa -Lots of bug fixes and improvements
2007-03-13 10:10:24 +01:00
include/gstreamer-0.10/gst/pbutils/descriptions.h
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
include/gstreamer-0.10/gst/pbutils/gstdiscoverer.h
include/gstreamer-0.10/gst/pbutils/gstpluginsbaseversion.h
update to 0.10.12 changes: -New API for on-demand plugin installation -Xv thread-safety and configuration enhancements -decodebin2 improvements -Support more raw audio format conversions -Improvements in Ogg support -AudioFilter base class ported to 0.10 -Fixes for subtitles -Latency/live-playback support for Alsa -Lots of bug fixes and improvements
2007-03-13 10:10:24 +01:00
include/gstreamer-0.10/gst/pbutils/install-plugins.h
include/gstreamer-0.10/gst/pbutils/missing-plugins.h
update to 0.10.17 changes: Handle newer Theora granule-pos semantics Introducing first alpha version playbin2 - the upcoming successor to playbin Fixes in playbin handling of stream-switching New API for uniform handling of raw-video format buffers Improvements for RTSP/RTP handling RIFF lib additions for VC-1 and AVC1 fourccs Many other bug-fixes and improvements
2008-02-01 21:44:16 +01:00
include/gstreamer-0.10/gst/pbutils/pbutils-enumtypes.h
update to 0.10.12 changes: -New API for on-demand plugin installation -Xv thread-safety and configuration enhancements -decodebin2 improvements -Support more raw audio format conversions -Improvements in Ogg support -AudioFilter base class ported to 0.10 -Fixes for subtitles -Latency/live-playback support for Alsa -Lots of bug fixes and improvements
2007-03-13 10:10:24 +01:00
include/gstreamer-0.10/gst/pbutils/pbutils.h
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
include/gstreamer-0.10/gst/riff/riff-ids.h
include/gstreamer-0.10/gst/riff/riff-media.h
include/gstreamer-0.10/gst/riff/riff-read.h
update to 0.10.11 changes: This release requires GStreamer core 0.10.11 or higher Improved buffering during network playback Support for reverse playback of Ogg/Vorbis/Theora Reverse playback support in the audio sinks The Ogg muxer now produces correct streams in all cases Support libvisual 0.4 Improvements in RTP support Support retagging of Vorbis files Introduction of the decodebin2 element Many other bug fixes and improvements
2006-12-11 13:46:08 +01:00
include/gstreamer-0.10/gst/rtp/gstbasertpaudiopayload.h
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
include/gstreamer-0.10/gst/rtp/gstbasertpdepayload.h
include/gstreamer-0.10/gst/rtp/gstbasertppayload.h
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
include/gstreamer-0.10/gst/rtp/gstrtcpbuffer.h
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
include/gstreamer-0.10/gst/rtp/gstrtpbuffer.h
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
include/gstreamer-0.10/gst/rtp/gstrtppayloads.h
Update to 0.10.14: Changes since 0.10.13: * 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 Bugs fixed since 0.10.13: * 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_...
2007-08-30 14:22:22 +02:00
include/gstreamer-0.10/gst/rtsp/gstrtsp-enumtypes.h
include/gstreamer-0.10/gst/rtsp/gstrtspbase64.h
include/gstreamer-0.10/gst/rtsp/gstrtspconnection.h
include/gstreamer-0.10/gst/rtsp/gstrtspdefs.h
include/gstreamer-0.10/gst/rtsp/gstrtspextension.h
include/gstreamer-0.10/gst/rtsp/gstrtspmessage.h
include/gstreamer-0.10/gst/rtsp/gstrtsprange.h
include/gstreamer-0.10/gst/rtsp/gstrtsptransport.h
include/gstreamer-0.10/gst/rtsp/gstrtspurl.h
include/gstreamer-0.10/gst/sdp/gstsdp.h
include/gstreamer-0.10/gst/sdp/gstsdpmessage.h
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
include/gstreamer-0.10/gst/tag/gsttagdemux.h
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
include/gstreamer-0.10/gst/tag/tag.h
include/gstreamer-0.10/gst/video/gstvideofilter.h
include/gstreamer-0.10/gst/video/gstvideosink.h
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
include/gstreamer-0.10/gst/video/video-enumtypes.h
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
include/gstreamer-0.10/gst/video/video.h
lib/gstreamer-0.10/libgstadder.la
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
lib/gstreamer-0.10/libgstapp.la
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
lib/gstreamer-0.10/libgstaudioconvert.la
lib/gstreamer-0.10/libgstaudiorate.la
lib/gstreamer-0.10/libgstaudioresample.la
lib/gstreamer-0.10/libgstaudiotestsrc.la
lib/gstreamer-0.10/libgstdecodebin.la
update to 0.10.11 changes: This release requires GStreamer core 0.10.11 or higher Improved buffering during network playback Support for reverse playback of Ogg/Vorbis/Theora Reverse playback support in the audio sinks The Ogg muxer now produces correct streams in all cases Support libvisual 0.4 Improvements in RTP support Support retagging of Vorbis files Introduction of the decodebin2 element Many other bug fixes and improvements
2006-12-11 13:46:08 +01:00
lib/gstreamer-0.10/libgstdecodebin2.la
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
lib/gstreamer-0.10/libgstffmpegcolorspace.la
update to 0.10.10 changes: -New elements: gdppay, gdpdepay -bugfixes -minor API additions
2006-10-12 15:30:36 +02:00
lib/gstreamer-0.10/libgstgdp.la
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
lib/gstreamer-0.10/libgstplaybin.la
lib/gstreamer-0.10/libgstsubparse.la
lib/gstreamer-0.10/libgsttcp.la
lib/gstreamer-0.10/libgsttypefindfunctions.la
lib/gstreamer-0.10/libgstvideorate.la
lib/gstreamer-0.10/libgstvideoscale.la
lib/gstreamer-0.10/libgstvideotestsrc.la
lib/gstreamer-0.10/libgstvolume.la
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
lib/libgstapp-0.10.la
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
lib/libgstaudio-0.10.la
Update to 0.10.2, ok jmmv@: Changes since 0.10.1: * ported gnomevfssink, cdparanoia * New library and base class: GstCddaBaseSrc * ported mixerutils.h * added 'sine-tab' waveform to audiotestsrc * added float audio to audiorate Bugs fixed since 0.10.1: * 324216 : [cdparanoia] missing patches from 0.8 * 324696 : [videotestsrc] does not start counting the time from zero... * 324900 : Problem compiling gst-plugins-base with Forte * 325984 : [playbin] cannot handle sources that produce raw audio/video * 325990 : patch videotestsrc for using glib types * 326601 : GstRingBuffer crashes with alaw/mulaw caps * 327114 : [theoradec] should post tags on the bus * 327216 : vorbisdec segfaults on certain queries
2006-01-25 20:52:47 +01:00
lib/libgstcdda-0.10.la
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
lib/libgstfft-0.10.la
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
lib/libgstinterfaces-0.10.la
lib/libgstnetbuffer-0.10.la
update to 0.10.12 changes: -New API for on-demand plugin installation -Xv thread-safety and configuration enhancements -decodebin2 improvements -Support more raw audio format conversions -Improvements in Ogg support -AudioFilter base class ported to 0.10 -Fixes for subtitles -Latency/live-playback support for Alsa -Lots of bug fixes and improvements
2007-03-13 10:10:24 +01:00
lib/libgstpbutils-0.10.la
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
lib/libgstriff-0.10.la
lib/libgstrtp-0.10.la
Update to 0.10.14: Changes since 0.10.13: * 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 Bugs fixed since 0.10.13: * 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_...
2007-08-30 14:22:22 +02:00
lib/libgstrtsp-0.10.la
lib/libgstsdp-0.10.la
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
lib/libgsttag-0.10.la
lib/libgstvideo-0.10.la
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
lib/pkgconfig/gstreamer-app-0.10.pc
update to 0.10.17 changes: Handle newer Theora granule-pos semantics Introducing first alpha version playbin2 - the upcoming successor to playbin Fixes in playbin handling of stream-switching New API for uniform handling of raw-video format buffers Improvements for RTSP/RTP handling RIFF lib additions for VC-1 and AVC1 fourccs Many other bug-fixes and improvements
2008-02-01 21:44:16 +01:00
lib/pkgconfig/gstreamer-audio-0.10.pc
lib/pkgconfig/gstreamer-cdda-0.10.pc
lib/pkgconfig/gstreamer-fft-0.10.pc
lib/pkgconfig/gstreamer-floatcast-0.10.pc
lib/pkgconfig/gstreamer-interfaces-0.10.pc
lib/pkgconfig/gstreamer-netbuffer-0.10.pc
lib/pkgconfig/gstreamer-pbutils-0.10.pc
Initial import of gst-plugins-0.10-base, version 0.10.1. This is an update for part of gst-plugins0.8 to the 0.10 series. Changes: Changes since 0.10.0: * Parallel installability with 0.8.x series * Threadsafe design and API * removed gst-launch-ext * Ported: ogmparse * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin Bugs fixed since 0.10.0: * 322347 : GstBaseRtpDepayload timestamps are wring * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered * 323878 : missing < string.h > inclusion (for memset & FD_ZERO) API added since 0.10.0: * GstAlsaMixer::device * GstAlsaMixer::device-name Bugs fixed since 0.9.7: * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags * 323017 : While(1) loop with sleep(0) in basertpdepayload.c Changes since 0.9.6: * Parallel installability with 0.8.x series * Threadsafe design and API * ximagesink and xvimagesink updates and interactive test * added pango * rename net to netbuffer library * rtp element renaming * stream selector fixes Bugs fixed since 0.9.6: * 319618 : [decodebin] some ogg videos don't play * 320644 : RTP packetizer does't set the packet timestamps correctly * 322388 : xvimagesink force-aspect-ratio=True always displays squar... * 322704 : oggdemux typefind list leak Changes since 0.9.5: * Parallel installability with 0.8.x series * Threadsafe design and API * lots of leak fixes * flicker-free and rewritten X sinks * fractional framerates * removed sinesrc, replaced by audiotestsrc Bugs fixed since 0.9.5: * 316442 : playbin should use autoaudiosink/autovideosink by default * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat... * 321164 : gstringbuffer stops working under load * 321426 : ximage plugin should be renamed to ximagesink * 321446 : sinesrc should be dropped in favour of audiotestsrc * 321451 : GstRtpBuffer: no way to create a sub buffer with only the... * 321816 : [API] xoverlay API to post prepare-xwindow-id message * 321894 : vorbisenc doesn't compile * 322117 : Rename libgsttagedit to libgsttag Changes since 0.9.4: * video caps now use a good range for framerate and w/h * oggdemux/oggmux improvements * playbin improvements Bugs fixed since 0.9.4: * 319110 : [PATCH] oggdemux chain finding is slow * 320058 : playbin of a jpeg over http does not work * 320923 : [volume] doesn't build on Solaris * 321011 : gstbasertpdepayload doesn't send the " new segment " event ... Changes since 0.9.3: * New element: audiotestsrc * typefind improvements * buffer-frames removed Changes since 0.9.2: * RTP base classes Bugs fixed since 0.9.2: * 313251 : ximagesink unused functions * 315159 : audioconvert lost 24 bit conversions in the rewrite
2006-01-18 16:41:03 +01:00
lib/pkgconfig/gstreamer-plugins-base-0.10.pc
update to 0.10.17 changes: Handle newer Theora granule-pos semantics Introducing first alpha version playbin2 - the upcoming successor to playbin Fixes in playbin handling of stream-switching New API for uniform handling of raw-video format buffers Improvements for RTSP/RTP handling RIFF lib additions for VC-1 and AVC1 fourccs Many other bug-fixes and improvements
2008-02-01 21:44:16 +01:00
lib/pkgconfig/gstreamer-riff-0.10.pc
lib/pkgconfig/gstreamer-rtp-0.10.pc
lib/pkgconfig/gstreamer-rtsp-0.10.pc
lib/pkgconfig/gstreamer-sdp-0.10.pc
lib/pkgconfig/gstreamer-tag-0.10.pc
lib/pkgconfig/gstreamer-video-0.10.pc
Update to 0.10.2, ok jmmv@: Changes since 0.10.1: * ported gnomevfssink, cdparanoia * New library and base class: GstCddaBaseSrc * ported mixerutils.h * added 'sine-tab' waveform to audiotestsrc * added float audio to audiorate Bugs fixed since 0.10.1: * 324216 : [cdparanoia] missing patches from 0.8 * 324696 : [videotestsrc] does not start counting the time from zero... * 324900 : Problem compiling gst-plugins-base with Forte * 325984 : [playbin] cannot handle sources that produce raw audio/video * 325990 : patch videotestsrc for using glib types * 326601 : GstRingBuffer crashes with alaw/mulaw caps * 327114 : [theoradec] should post tags on the bus * 327216 : vorbisdec segfaults on certain queries
2006-01-25 20:52:47 +01:00
man/man1/gst-visualise-0.10.1
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/annotation-glossary.html
update to 0.10.26 changes: -many fixes to playbin2 -libgsttag: add utility functions for ISO-639 language codes and tags -theora: port to 'new' theora 1.0 API -more fixes and improvements
2010-02-16 14:14:31 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/api-index-deprecated.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/api-index-full.html
update to 0.10.28 changes: -videorate: Improve upstream negotiation -ffmpegcolorspace: add conversions from all ARGB formats to AYUV and back -vorbisdec: also support ivorbis tremor decoder -misc fixes
2010-03-12 21:31:03 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/compiling.html
update to 0.10.21 changes: Continue playbin2 development Ogg improvements - CELT support, skeleton fixes DVD subpicture support Improved audio dithering random number generator xvimagesink/ximagesink fixes Vorbis encoding and decoding fixes Recognise Kate subtitle streams Many bug-fixes and enhancements
2008-10-16 18:22:49 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/compiling.sgml
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-0.10.devhelp
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-0.10.devhelp2
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-appsink.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-appsrc.html
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstaudio.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstaudioclock.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstaudiofilter.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstaudiomixerutils.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstaudiosink.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstaudiosrc.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstbaseaudiosink.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstbaseaudiosrc.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstbasertpaudiopayload.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstbasertpdepayload.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstbasertppayload.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstcddabasesrc.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstcolorbalance.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstcolorbalancechannel.html
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstdiscoverer.html
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstfft.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstfftf32.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstfftf64.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstffts16.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstffts32.html
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstfloatcast.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstmixer.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstmixeroptions.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstmixertrack.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstmultichannel.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstnavigation.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstnetbuffer.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstpbutils.html
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstpbutilscodecutils.html
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstpbutilsdescriptions.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstpbutilsinstallplugins.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstpbutilsmissingplugins.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstpropertyprobe.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstriff.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstringbuffer.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtcpbuffer.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtpbuffer.html
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtppayloads.html
Update to 0.10.14: Changes since 0.10.13: * 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 Bugs fixed since 0.10.13: * 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_...
2007-08-30 14:22:22 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtspbase64.html
update to 0.10.30 changes: -Use Orc for SIMD and other optimisations, and remove liboil dependency -fixes, minor feature extensions
2010-07-22 14:57:15 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtspconnection.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtspdefs.html
update to 0.10.29 changes: * video: add support for color-matrix and chroma-site fields in video caps and selected elements * video: Add support for 8-bit and 16-bit grayscale formats * typefinding: add AAC profile, level, channels and rate to ADTS caps * tags: add basic xmp metadata support * gio, gnomevfs: invert ranks of gio and gnomevfs elements: gio is prefered now, gnomevfs has been deprecated * riff: add mapping for On2 VP62 and VP7 and add some more MPEG4 fourccs * playsink: Don't fail if there are subtitles and audio but no video * oggdemux: map old FLAC mapping correctly * alsa: handle disappearing of sound device whilst in use more gracefully * playbin: Only unref the volume element on dispose and when a new audio sink is set * build: build plugin, example and libs directories in parallel if make -jN is used * uridecodebin/playbin2: we can handle avi in download mode too * rtsp: handle closed POST socket in tunneling, ignore unparsable ranges, allow for more ipv6 addresses * audiopayload: add property to control packet duration
2010-05-14 18:28:57 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtspextension.html
update to 0.10.30 changes: -Use Orc for SIMD and other optimisations, and remove liboil dependency -fixes, minor feature extensions
2010-07-22 14:57:15 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtspmessage.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtsprange.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtsptransport.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstrtspurl.html
Update to 0.10.14: Changes since 0.10.13: * 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 Bugs fixed since 0.10.13: * 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_...
2007-08-30 14:22:22 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstsdpmessage.html
update to 0.10.25 changes: Add per-stream volume controls Theora 1.0 and Y444 and Y42B format support Improve audio capture timing GObject introspection support Improve audio output startup RTSP improvements Use pango-cairo instead of pangoft2 Allow cdda://(device#)?track URI scheme in cddabasesrc Support interlaced content in videoscale and ffmpegcolorspacee Many other bug fixes and improvements
2009-10-21 18:45:31 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gststreamvolume.html
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gsttag.html
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gsttagdemux.html
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gsttagid3.html
update to 0.10.26 changes: -many fixes to playbin2 -libgsttag: add utility functions for ISO-639 language codes and tags -theora: port to 'new' theora 1.0 API -more fixes and improvements
2010-02-16 14:14:31 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gsttaglanguagecodes.html
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gsttagvorbis.html
update to 0.10.29 changes: * video: add support for color-matrix and chroma-site fields in video caps and selected elements * video: Add support for 8-bit and 16-bit grayscale formats * typefinding: add AAC profile, level, channels and rate to ADTS caps * tags: add basic xmp metadata support * gio, gnomevfs: invert ranks of gio and gnomevfs elements: gio is prefered now, gnomevfs has been deprecated * riff: add mapping for On2 VP62 and VP7 and add some more MPEG4 fourccs * playsink: Don't fail if there are subtitles and audio but no video * oggdemux: map old FLAC mapping correctly * alsa: handle disappearing of sound device whilst in use more gracefully * playbin: Only unref the volume element on dispose and when a new audio sink is set * build: build plugin, example and libs directories in parallel if make -jN is used * uridecodebin/playbin2: we can handle avi in download mode too * rtsp: handle closed POST socket in tunneling, ignore unparsable ranges, allow for more ipv6 addresses * audiopayload: add property to control packet duration
2010-05-14 18:28:57 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gsttagxmp.html
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gsttuner.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gsttunerchannel.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gsttunernorm.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstvideo.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstvideofilter.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstvideoorientation.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstvideosink.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gst-plugins-base-libs-gstxoverlay.html
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-app.html
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-audio.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-base-utils.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-cdda.html
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-ffft.html
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-floatcast.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-interfaces.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-libs-hierarchy.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-netbuffer.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-plugins-base.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-riff.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-rtp.html
Update to 0.10.14: Changes since 0.10.13: * 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 Bugs fixed since 0.10.13: * 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_...
2007-08-30 14:22:22 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-rtsp.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-sdp.html
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-tag.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/gstreamer-video.html
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/home.png
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/index.html
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/index.sgml
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/left.png
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/right.png
Update to 0.10.13: Changes since 0.10.12: * Many fixes and improvements * RTP and RTCP support improved API additions since 0.10.12: * add gst_tag_freeform_string_to_utf8() * GstRTPBuffer::gst_rtp_buffer_default_clock_rate() * GstBaseAudioSink::slave-method property * add "min-ptime" property to RTP base audio payloader * gst_base_rtp_audio_payload_push() * gst_base_rtp_audio_payload_get_adapter() * GstMixerTrack::untranslated-label property
2007-06-08 14:58:11 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/style.css
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-libs-0.10/up.png
update to 0.10.21 changes: Continue playbin2 development Ogg improvements - CELT support, skeleton fixes DVD subpicture support Improved audio dithering random number generator xvimagesink/ximagesink fixes Vorbis encoding and decoding fixes Recognise Kate subtitle streams Many bug-fixes and enhancements
2008-10-16 18:22:49 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/GstPlayBaseBin.html
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/ch01.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/ch02.html
update to 0.10.21 changes: Continue playbin2 development Ogg improvements - CELT support, skeleton fixes DVD subpicture support Improved audio dithering random number generator xvimagesink/ximagesink fixes Vorbis encoding and decoding fixes Recognise Kate subtitle streams Many bug-fixes and enhancements
2008-10-16 18:22:49 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/ch03.html
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-0.10.devhelp
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-0.10.devhelp2
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-adder.html
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-alsamixer.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-alsasink.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-alsasrc.html
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-appsink.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-appsrc.html
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-audioconvert.html
update to 0.10.26 changes: -many fixes to playbin2 -libgsttag: add utility functions for ISO-639 language codes and tags -theora: port to 'new' theora 1.0 API -more fixes and improvements
2010-02-16 14:14:31 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-audiorate.html
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-audioresample.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-audiotestsrc.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-cdparanoiasrc.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-clockoverlay.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-decodebin.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-decodebin2.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-ffmpegcolorspace.html
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-gdpdepay.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-gdppay.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-giosink.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-giosrc.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-giostreamsink.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-giostreamsrc.html
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-gnomevfssink.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-gnomevfssrc.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-multifdsink.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-oggdemux.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-oggmux.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-playbin.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-playbin2.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-adder.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-alsa.html
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-app.html
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-audioconvert.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-audiorate.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-audioresample.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-audiotestsrc.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-cdparanoia.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-decodebin.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-ffmpegcolorspace.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-gdp.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-gio.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-gnomevfs.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-libvisual.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-ogg.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-pango.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-playback.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-subparse.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-tcp.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-theora.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-typefindfunctions.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-uridecodebin.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-video4linux.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-videorate.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-videoscale.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-videotestsrc.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-volume.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-vorbis.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-ximagesink.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-plugin-xvimagesink.html
update to 0.10.26 changes: -many fixes to playbin2 -libgsttag: add utility functions for ISO-639 language codes and tags -theora: port to 'new' theora 1.0 API -more fixes and improvements
2010-02-16 14:14:31 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-subtitleoverlay.html
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-tcpclientsink.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-tcpclientsrc.html
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-tcpserversink.html
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-tcpserversrc.html
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-textoverlay.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-textrender.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-theoradec.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-theoraenc.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-theoraparse.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-timeoverlay.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-uridecodebin.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-videorate.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-videoscale.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-videotestsrc.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-volume.html
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-vorbisdec.html
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-vorbisenc.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-vorbisparse.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-vorbistag.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-ximagesink.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/gst-plugins-base-plugins-xvimagesink.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/home.png
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/index.html
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/index.sgml
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/left.png
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/right.png
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/style.css
share/doc/gst-plugins-0.10/gst-plugins-base-plugins-0.10/up.png
Strip ${PKGLOCALEDIR} from PLISTs of packages that already obey PKGLOCALEDIR and which install their locale files directly under ${PREFIX}/${PKGLOCALEDIR} and sort the PLIST file entries. From now on, pkgsrc/mk/plist/plist-locale.awk will automatically handle transforming the PLIST to refer to the correct locale directory.
2006-04-17 09:07:11 +02:00
share/locale/af/LC_MESSAGES/gst-plugins-base-0.10.mo
share/locale/az/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
share/locale/bg/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.17 changes: Handle newer Theora granule-pos semantics Introducing first alpha version playbin2 - the upcoming successor to playbin Fixes in playbin handling of stream-switching New API for uniform handling of raw-video format buffers Improvements for RTSP/RTP handling RIFF lib additions for VC-1 and AVC1 fourccs Many other bug-fixes and improvements
2008-02-01 21:44:16 +01:00
share/locale/ca/LC_MESSAGES/gst-plugins-base-0.10.mo
Strip ${PKGLOCALEDIR} from PLISTs of packages that already obey PKGLOCALEDIR and which install their locale files directly under ${PREFIX}/${PKGLOCALEDIR} and sort the PLIST file entries. From now on, pkgsrc/mk/plist/plist-locale.awk will automatically handle transforming the PLIST to refer to the correct locale directory.
2006-04-17 09:07:11 +02:00
share/locale/cs/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
share/locale/da/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.12 changes: -New API for on-demand plugin installation -Xv thread-safety and configuration enhancements -decodebin2 improvements -Support more raw audio format conversions -Improvements in Ogg support -AudioFilter base class ported to 0.10 -Fixes for subtitles -Latency/live-playback support for Alsa -Lots of bug fixes and improvements
2007-03-13 10:10:24 +01:00
share/locale/de/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
share/locale/el/LC_MESSAGES/gst-plugins-base-0.10.mo
Strip ${PKGLOCALEDIR} from PLISTs of packages that already obey PKGLOCALEDIR and which install their locale files directly under ${PREFIX}/${PKGLOCALEDIR} and sort the PLIST file entries. From now on, pkgsrc/mk/plist/plist-locale.awk will automatically handle transforming the PLIST to refer to the correct locale directory.
2006-04-17 09:07:11 +02:00
share/locale/en_GB/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
share/locale/es/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.25 changes: Add per-stream volume controls Theora 1.0 and Y444 and Y42B format support Improve audio capture timing GObject introspection support Improve audio output startup RTSP improvements Use pango-cairo instead of pangoft2 Allow cdda://(device#)?track URI scheme in cddabasesrc Support interlaced content in videoscale and ffmpegcolorspacee Many other bug fixes and improvements
2009-10-21 18:45:31 +02:00
share/locale/eu/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
share/locale/fi/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/locale/fr/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
share/locale/gl/LC_MESSAGES/gst-plugins-base-0.10.mo
Strip ${PKGLOCALEDIR} from PLISTs of packages that already obey PKGLOCALEDIR and which install their locale files directly under ${PREFIX}/${PKGLOCALEDIR} and sort the PLIST file entries. From now on, pkgsrc/mk/plist/plist-locale.awk will automatically handle transforming the PLIST to refer to the correct locale directory.
2006-04-17 09:07:11 +02:00
share/locale/hu/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.21 changes: Continue playbin2 development Ogg improvements - CELT support, skeleton fixes DVD subpicture support Improved audio dithering random number generator xvimagesink/ximagesink fixes Vorbis encoding and decoding fixes Recognise Kate subtitle streams Many bug-fixes and enhancements
2008-10-16 18:22:49 +02:00
share/locale/id/LC_MESSAGES/gst-plugins-base-0.10.mo
Strip ${PKGLOCALEDIR} from PLISTs of packages that already obey PKGLOCALEDIR and which install their locale files directly under ${PREFIX}/${PKGLOCALEDIR} and sort the PLIST file entries. From now on, pkgsrc/mk/plist/plist-locale.awk will automatically handle transforming the PLIST to refer to the correct locale directory.
2006-04-17 09:07:11 +02:00
share/locale/it/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.22 changes: Require gettext 0.17 Replace audioresample with speexresample from -bad Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 Move libgstapp and elements from -bad Support color-key setting and probing for Xv properties Improve typefinding for various formats Extend audio sinks for pull-mode operation Support for more subtitle formats More development on decode2bin and playbin2 RTP and SDP fixes Many bug fixes and improvements
2009-01-26 11:39:01 +01:00
share/locale/ja/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.19 changes: -Experimental GIO plugin -Continued playbin2 development -RTP fixes -Various other bug-fixes and improvements
2008-04-10 14:00:45 +02:00
share/locale/lt/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.24 changes: Recognise Kate subpicture subtitles Support progressive download in playbin2 GIO improvements Add buffer-list support in appsink Add gaussian-noise mode to audiotestsrc bump cdparanoia req to 0.10.2 and improve caching Improve audio source base class Add frame-by-frame stepping and examples Extend stream-probing in decodebin2 Many RTSP improvements support for PGS subpictures adder improvements Add Y444, v210, v216 formats implement preset interface in vorbisenc, theoraenc, oggmux Improve libvisual visualisation timestamp tracking playbin2 enhancements: custom audiosink, subpictures, cdda Improvements in textrender Support raw YUV 4:2:2 and SIREN in RIFF Add 4:2:2 and 4:4:4 support to theoradec Many other bug-fixes and improvements
2009-08-17 19:45:52 +02:00
share/locale/lv/LC_MESSAGES/gst-plugins-base-0.10.mo
Strip ${PKGLOCALEDIR} from PLISTs of packages that already obey PKGLOCALEDIR and which install their locale files directly under ${PREFIX}/${PKGLOCALEDIR} and sort the PLIST file entries. From now on, pkgsrc/mk/plist/plist-locale.awk will automatically handle transforming the PLIST to refer to the correct locale directory.
2006-04-17 09:07:11 +02:00
share/locale/nb/LC_MESSAGES/gst-plugins-base-0.10.mo
share/locale/nl/LC_MESSAGES/gst-plugins-base-0.10.mo
share/locale/or/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
share/locale/pl/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.21 changes: Continue playbin2 development Ogg improvements - CELT support, skeleton fixes DVD subpicture support Improved audio dithering random number generator xvimagesink/ximagesink fixes Vorbis encoding and decoding fixes Recognise Kate subtitle streams Many bug-fixes and enhancements
2008-10-16 18:22:49 +02:00
share/locale/pt_BR/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.31 changes: many fixes and improvements - see the Changelog
2010-12-03 19:05:58 +01:00
share/locale/ro/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.20 changes: RTP improvements Support digest auth for RTSP Additional documentation Support DSCP QoS in multifdsink Add NV12/NV21 video buffer layouts Video scaling now bilinear by default Support more than 8 channels in audio conversions Channel mapping fixes for audioconvert Improve tmplayer and sami subtitle support Support 1x1 pixel buffers for videoscale Typefinding improvements for MPEG2, musepack Ogg/Dirac mapping updated in oggmux Fixes in ogg demuxing audiosink synchronisation and slaving fixes Support muting of the audio in playbin by selecting -1 as the audio stream Work done on playbin2 and uridecodebin Improvements in the experimental GIO plugin decodebin fixes Handle GAP buffers in some places Various other leak and bug-fixes
2008-07-15 12:09:40 +02:00
share/locale/ru/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.17 changes: Handle newer Theora granule-pos semantics Introducing first alpha version playbin2 - the upcoming successor to playbin Fixes in playbin handling of stream-switching New API for uniform handling of raw-video format buffers Improvements for RTSP/RTP handling RIFF lib additions for VC-1 and AVC1 fourccs Many other bug-fixes and improvements
2008-02-01 21:44:16 +01:00
share/locale/sk/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.30 changes: -Use Orc for SIMD and other optimisations, and remove liboil dependency -fixes, minor feature extensions
2010-07-22 14:57:15 +02:00
share/locale/sl/LC_MESSAGES/gst-plugins-base-0.10.mo
Strip ${PKGLOCALEDIR} from PLISTs of packages that already obey PKGLOCALEDIR and which install their locale files directly under ${PREFIX}/${PKGLOCALEDIR} and sort the PLIST file entries. From now on, pkgsrc/mk/plist/plist-locale.awk will automatically handle transforming the PLIST to refer to the correct locale directory.
2006-04-17 09:07:11 +02:00
share/locale/sq/LC_MESSAGES/gst-plugins-base-0.10.mo
share/locale/sr/LC_MESSAGES/gst-plugins-base-0.10.mo
share/locale/sv/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.24 changes: Recognise Kate subpicture subtitles Support progressive download in playbin2 GIO improvements Add buffer-list support in appsink Add gaussian-noise mode to audiotestsrc bump cdparanoia req to 0.10.2 and improve caching Improve audio source base class Add frame-by-frame stepping and examples Extend stream-probing in decodebin2 Many RTSP improvements support for PGS subpictures adder improvements Add Y444, v210, v216 formats implement preset interface in vorbisenc, theoraenc, oggmux Improve libvisual visualisation timestamp tracking playbin2 enhancements: custom audiosink, subpictures, cdda Improvements in textrender Support raw YUV 4:2:2 and SIREN in RIFF Add 4:2:2 and 4:4:4 support to theoradec Many other bug-fixes and improvements
2009-08-17 19:45:52 +02:00
share/locale/tr/LC_MESSAGES/gst-plugins-base-0.10.mo
Strip ${PKGLOCALEDIR} from PLISTs of packages that already obey PKGLOCALEDIR and which install their locale files directly under ${PREFIX}/${PKGLOCALEDIR} and sort the PLIST file entries. From now on, pkgsrc/mk/plist/plist-locale.awk will automatically handle transforming the PLIST to refer to the correct locale directory.
2006-04-17 09:07:11 +02:00
share/locale/uk/LC_MESSAGES/gst-plugins-base-0.10.mo
share/locale/vi/LC_MESSAGES/gst-plugins-base-0.10.mo
update to 0.10.15 changes: 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
2007-11-22 21:28:44 +01:00
share/locale/zh_CN/LC_MESSAGES/gst-plugins-base-0.10.mo
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