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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