RELEASE: GStreamer Base Plug-ins 0.10.23 'Emergency de-stress call'
- From: thaytan noraisin net
- To: gstreamer-devel lists sourceforge net, gstreamer-announce lists sourceforge net, kde-multimedia kde org, gnome-multimedia gnome org
- Subject: RELEASE: GStreamer Base Plug-ins 0.10.23 'Emergency de-stress call'
- Date: Mon, 11 May 2009 00:41:39 +0100 (IST)
This mail announces the release of GStreamer Base Plug-ins 0.10.23 'Emergency de-stress call'.
GStreamer Base Plug-ins is a well-groomed and well-maintained collection of
GStreamer plug-ins and elements, spanning the range of possible types of
elements one would want to write for GStreamer. It also contains helper
libraries and base classes useful for writing elements.
A wide range of video and audio decoders, encoders, and filters are included.
For more information, see http://gstreamer.freedesktop.org/modules/gst-plugins-base.html
To file bugs, go to http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-plugins-base
Release notes for GStreamer Base Plug-ins 0.10.23 "Emergency de-stress call"
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
* New navigation API to support DVD playback
* playbin2 improvements
* RTSP extensions to allow extra headers and options
* Replace audioresampler with speexresample based code
* Support interlacing flags in the gstvideo library
* Support new RIFF formats
* Improve typefinding
* Support more frame formats in videoscale
* Many other bug-fixes and improvements
Bugs fixed in this release
* 577637 : [playbin2] expose temp-location property
* 580120 : [playbin2] unit test fails
* 478512 : [alsamixer] volume control slider not working
* 574962 : rhythmbox crash in flac_type_find
* 564139 : Documentation of TCP plugins
* 577436 : xvimagesink should use xcontext- > depth and not count bits...
* 350311 : [playbin2] support for subpicture subtitles
* 378094 : Enable pango elements to handle UYVY
* 543591 : Gnonlin can not play theora streams
* 553295 : [riff] fuzzed AVI file causes segfault
* 565105 : Gstreamer does not change from READY back to PAUSED in sa...
* 565777 : [riff] unrecognised video fourcc 0x10000002 for mpeg2 in avi
* 566661 : [typefind] Fall back to file extension using uri query
* 567255 : [riff] doesn't detect codec_id 0x706d as AAC (amongst other)
* 567636 : [pbutils] Missing plugins code shouldn't ask for the same...
* 567740 : bogus warning in decodebin2?
* 568482 : linking problems in gst-plugins-base
* 569655 : [ffmpegcolorspace] Add UYVY422 to GRAY8 conversion function
* 570142 : Documentation is broken for uridecodebin
* 570356 : aac typefinder failure
* 570768 : [ximagesink] wrong mouse pointer position if output windo...
* 570832 : Add flags to enhance mixer interfaces
* 571009 : [tagdemux] WMA file with id3v2 tag causes assertion to fail
* 571147 : [ffmpegcolorspace/videotestsrc] Add support for packed/pl...
* 572577 : [playbin2] deadlock on shutdown
* 572872 : [ffmpegcolorspace] Add YVYU colorspace
* 572993 : [subparse] broken libregex dependency on Windows
* 573165 : Generate additional export files for gstreamer app plugin
* 573528 : Wrong format modifier in gstgiobasesink.c
* 573529 : In gstrtspconnection.c some functions are called with wro...
* 574293 : [decodebin2] deadlock on shutdown
* 574319 : Missing HAVE_PROCESS_H in win32/common/config.h
* 574447 : gstadder.c: line 904: error C2036: 'gpointer' : unknown size
* 574939 : [typefinding] flac typefinder mis-typefinds PDFs as flac ...
* 575550 : srt subtitle file keeps playbin2 from playing
* 575638 : kissfft copyright
* 575649 : [oggdemux] duration query in time format returns true wit...
* 576019 : On Windows queue2 can't write files longer than 2-4 GiB, ...
* 576142 : [vorbisenc] Non-header output buffers have NULL caps
* 576180 : [playbin2] Uses unref'd audiosink volume if using gconfau...
* 576586 : [alsamixer] gnome-sound-properties freeze
* 577054 : [videoscale] Not valgrind clean
* 577709 : Review new navigation API
* 577827 : [appsink] Have appsink new_buffer-callback return GstFlow...
* 578583 : [PATCH] multifdsink doesn't handle sync-method=latest-key...
* 578656 : Implement upstream GstForceKeyUnit events in theoraenc
* 579129 : pkgconfig: appsrc/appsink can not be linked to uninstalled
* 579130 : app: expose trivial type macros
* 579192 : gst_rtcp_packet_get_type should not assert on packet content
* 579203 : baseaudiosink: unparenting the ringbuffer in NULL causes ...
* 579267 : [rtspconnection] g_async_queue_new_full() is GLib-2.16 AP...
* 579463 : [cddabasesrc] [cdparanoiasrc] no longer emits discid
* 579668 : audioresample fails to build with --disable-gst-debug
* 579734 : [playbin] raw_decoding_mode seems to be set unconditionally
* 579912 : [decodebin2] multiqueue is too small in time (interleave ...
* 580470 : [audioresample] causes pipelines to go out of sync and be...
* 580952 : [audioresample] bad quality/pops compared to plughw
* 581727 : [playbin2] make playsink go to PAUSED async
* 569682 : playbin2 leaks request pad from input selector
* 580020 : [vorbisenc] causes buffers to be out of segment if new se...
* 562794 : rtspsrc fails to create a socket on Win32 sometimes.
* 567396 : playbin2: DECODE_BIN_LOCK occasionally called twice withi...
* 567982 : " queued_bytes " field isn't updated while flushing the que...
* 571299 : [appsink] Handoff callback API
* 574443 : rtsp win32 - forgotten variable
* 574516 : [typefind] add typefinder for photoshop .psd files
* 574964 : gst_app_src_end_of_stream(), mutex on error return
* 575256 : rtspsrc fails to resolve hostnames
* 575588 : decodebin2 deadlock
* 576187 : [playbin2] Stalls video sink when disabling subtitles in ...
* 576188 : [playbin2] Reusing a playbin2 instance with visualization...
* 576190 : [playbin2] Deadlock when reusing playbin2 after an error
* 577288 : " Internal playbin error " when seeking to the end of files
* 577610 : RTCP feedback messages support in GstRTCPPacket
* 577794 : [playbin2] leaks elements set through properties
* 578118 : [multifdsink] add option to not resend the streamheader w...
* 578506 : Pipeline with alsasrc and alsasink cannot change state ba...
* 578942 : Missing RTSP headers related to Windows Media extension.
* 580271 : videorate: fails to clear discont flag on duplicated buffers
* 580649 : uridecodebin: bug on documentation published in website
API changed in this release
- API additions:
* GstRTSP::gst_rtsp_options_as_text()
* GstRTSPMessage::gst_rtsp_message_take_header()
* GstRTSPRange::gst_rtsp_range_to_string()
* New Navigation interface commands, queries and messages
* gst_rtsp_channel_new()
* gst_rtsp_channel_unref()
* gst_rtsp_channel_attach()
* gst_rtsp_channel_queue_message()
* gst_rtsp_connection_accept()
* GstAppSink::gst_app_sink_set_callbacks()
* GST_VIDEO_FORMAT_YVYU,GST_VIDEO_BUFFER_TFF,GST_VIDEO_BUFFER_RFF,GST_VIDEO_BUFFER_ONEFIELD
* GST_MIXER_FLAG_HAS_WHITELIST,GST_MIXER_FLAG_GROUPING,GST_MIXER_TRACK_NO_RECORD,GST_MIXER_TRACK_NO_MUTE,GST_MIXER_TRACK_WHITELIST
* GstAppSrc::emit-signals
* GstAppSrc::gst_app_src_set_emit_signals()
* GstAppSrc::gst_app_src_get_emit_signals()
* GstAppSrc::gst_app_src_set_callbacks()
* RTSP::gst_rtsp_connection_get_url()
* GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
* RTSP:gst_rtsp_connection_set_tunneled()
* RTSP:gst_rtsp_connection_is_tunneled()
* RTSP::gst_rtsp_connection_set_ip()
* RTSP::gst_rtsp_connection_get_tunnelid()
* RTSP::gst_rtsp_connection_do_tunnel()
* RTSP::gst_rtsp_watch_reset()
Download
You can find source releases of gst-plugins-base in the download directory:
http://gstreamer.freedesktop.org/src/gst-plugins-base/
GStreamer Homepage
More details can be found on the project's website:
http://gstreamer.freedesktop.org/
Support and Bugs
We use GNOME's bugzilla for bug reports and feature requests:
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer
Developers
GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned from there.
Interested developers of the core library, plug-ins, and applications should
subscribe to the gstreamer-devel list. If there is sufficient interest we
will create more lists as necessary.
Applications
Contributors to this release
* Andy Wingo
* Antoine Tremblay
* Benjamin Gaignard
* Benjamin M. Schwartz
* Brian Cameron
* Christian Schaller
* David Flynn
* David Schleef
* Edward Hervey
* Felipe Contreras
* Garret D'Amore
* Hannes Bistry
* Jan Schmidt
* Jan Urbanski
* Johann Prieur
* Jonas Danielsson
* Jonathan Matthew
* Josep Torra
* Julien Moutte
* Luca Ognibene
* Mark Nauwelaerts
* Martin Samuelsson
* Michael Smith
* Olivier Crete
* Peter Kjellerstedt
* René Stadler
* Sebastian Dröge
* Stefan Kost
* Tim-Philipp Müller
* Tomas Hoger
* Wim Taymans
* Zaheer Merali
* Zeeshan Ali
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