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
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
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
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_...
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
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
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
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.
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
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
