|
 |
Release notes for
GStreamer Base Plug-ins 0.10.22
"hidey hidey hidey ho"
The GStreamer team is proud to announce a new release
in the 0.10.x stable series of the
GStreamer Base Plug-ins.
The 0.10.x series is a stable series targeted at end users.
It is not API or ABI compatible with the stable 0.8.x series.
It is, however, parallel installable with the 0.8.x series.
This module contains a set of reference plugins, base classes for other
plugins, and helper libraries.
This module is kept up-to-date together with the core developments. Element
writers should look at the elements in this module as a reference for
their development.
This module contains elements for, among others:
- device plugins: x(v)imagesink, alsa, v4lsrc, cdparanoia
- containers: ogg
- codecs: vorbis, theora
- text: textoverlay, subparse
- sources: audiotestsrc, videotestsrc, gnomevfssrc
- network: tcp
- typefind
- audio processing: audioconvert, adder, audiorate, audioscale, volume
- visualisation: libvisual
- video processing: ffmpegcolorspace
- aggregate elements: decodebin, playbin
Other modules containing plug-ins are:
- gst-plugins-good
- contains a set of well-supported plug-ins under our preferred license
- gst-plugins-ugly
- contains a set of well-supported plug-ins, but might pose problems for
distributors
- gst-plugins-bad
- contains a set of less supported plug-ins that haven't passed the
rigorous quality testing we expect
Features of this release
- 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
Bugs fixed in this release
-
562163
: theoraenc likely ignoring segments
-
562258
: rtspsrc element takes long time to error out if the addre...
-
561789
: [volume] deadlocks with a controller attached
-
554533
: [xvimagesink] allow setting colorkey if possible
-
567511
: colorkey in xvimagesink gets reset when element is reused
-
116051
: libresample doesn't handle > factor of 2 rate conversion
-
346218
: [audioresample] doesn't do anti aliasing
-
385061
: [audioresample?] investigate high CPU usage
-
456788
: [subparse] can't handle UTF-16 charset encoded subtitle.
-
525807
: [vorbisenc] vorbisenc has problems with a gnlsource that ...
-
546955
: gstoggmux EOS handling issue
-
549417
: [audioresample] unit test fails on 64bit linux
-
549510
: audioresample doesn't negotiate ideal caps
-
552237
: UTF-16 srt confuses gstreamer, misdetected as mp3
-
552559
: Implementation of SLAVE_SKEW in baseaudiosrc
-
552569
: audioresample producing strange sized buffers
-
552801
: audioconvert can overflow with big audio buffers
-
554879
: Add ability to specify format for date/time display in Gs...
-
555257
: Doesn't display srt subtitles saved with BOM
-
555319
: add FFV1 fourcc to riff-media
-
555607
: subrip subtitles typefind too strict
-
555699
: [PATCH] theoradec: prefer container's pixel aspect ratio ...
-
556025
: build failure in tests/icles
-
556066
: Last byte of FLAC image buffer chopped off
-
557365
: subparse check fails
-
558124
: [PLUGIN-MOVE] Move speexresample as audioresample2 to -base
-
559111
: ALSA sink hangs on USB audio device unplug while playing
-
559478
: does not play windows media streams correctly
-
559567
: `gst_base_audio_sink_sync_latency' should call `gst_base_...
-
561436
: videorate element add image/jpeg to caps template
-
561734
: playbin2 additions
-
561780
: Playbin2 should work without volume too
-
561924
: oggdemux hangs when given corrupt input via non-seekable ...
-
562270
: build without gdk fails
-
563143
: ximagesink/xvimagesink : _alloc_buffer returns non-clean ...
-
563174
: Implement gst_rtcp_packet_remove
-
563508
: [rgvolume] Unit test fails with passthrough assertions
-
563718
: Theora check out of date
-
563904
: GNOME Goal: Clean up GLib and GTK+ includes
-
564098
: MS Word files are recognised as audio/mpeg and OSX's .DS_...
-
564139
: Documentation of TCP plugins
-
564200
: GstBaseAudioSink should register its enums and have corre...
-
564206
: GstBaseAudioSrc should register its enum and have corresp...
-
564421
: Move appsrc/appsink to -base
-
564929
: Audiosink blocks if setcaps called while playing
-
566586
: playbin2 test7.c fails after two songs
-
566750
: [appsrc/sink] add padding, move private data to private s...
-
566761
: [gstapp] No pkg-config file
-
566837
: gst_cdda_base_src_mode_get_type() is not public from < gst...
-
566875
: [gnomevfs] Add dependency for the GnomeVFS modules
-
566876
: [gio] Add dependency for the modules dir
-
567027
: Add GType for GstRTSPUrl for bindings
-
567168
: appsink is using the wrong signal slot for the pull-buffe...
-
567960
: [tagdemux] Doesn't forward unknown events upstream
-
500833
: [FFT] Struct alignment issues on sparc
-
552199
: Parsing SDP file with multicast address fails
-
558553
: [riff] gst_riff_create_video_caps not recognizing certain...
-
564896
: gst_netaddress_get_ip[46]_address should check for correc...
-
566341
: Some Ogg Theora files don't finished at seek at the end
-
566654
: playbin2: does not come back from NULL after switching UR...
-
566723
: GstAudioClock's new function may better use const gchar* ...
API changes
- API additions
- clockoverlay::time-format
- GstRingBuffer:gst_ring_buffer_activate()
- GstRingBuffer:gst_ring_buffer_is_active()
- GstRingBuffer:gst_ring_buffer_convert()
- Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API
- gst_netaddress_get_address_bytes()
- gst_netaddress_set_address_bytes()
Download
You can find source releases of gst-plugins-base in the
gst-plugins-base download directory.
The git repository and details how to clone it can be found at
git.freedesktop.org
.
Homepage
The project's website is https://gstreamer.freedesktop.org.
Support and Bugs
We use GNOME's bugzilla for
bug reports and feature requests.
Please submit patches via bugzilla as well.
For help and support, please subscribe to and send questions to the
gstreamer-devel mailing list (see below for details).
Find us on IRC at #gstreamer.
Developers
Git is hosted on git.freedesktop.org. You can
browse the gst-plugins-base repository.
All code is in Git and can be checked out from there.
Interested developers of the core library, plugins, and applications should
subscribe to the gstreamer-devel list.
Applications
Contributors to this release
- Alessandro Decina
- Andrew Feren
- Andy Wingo
- Christian Schaller
- Cygwin Ports maintainer
- Damien Lespiau
- Daniel Drake
- David Schleef
- Edward Hervey
- Guillaume Emont
- Håvard Graff
- Jan Gerber
- Jan Schmidt
- Jonathan Matthew
- Jonathan Rosser
- José Alburquerque
- Julien Moutte
- Klaas
- Luis Menina
- Mark Nauwelaerts
- Matthias Kretz
- Michael Smith
- Nick Haddad
- Olivier Crete
- Pavel Zeldin
- Robin Stocker
- Sebastian Dröge
- Stefan Kost
- Tero Saarni
- Thomas Vander Stichele
- Tim-Philipp Müller
- Wim Taymans
- xavierb at gmail dot com
- 이문형
|
|
