RELEASE: GStreamer Base Plug-ins 0.10.22 'Hidey Hidey Hidey Ho'

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 changed in this release
     
- 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 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

CVS is hosted on cvs.freedesktop.org.
All code is in CVS and can be checked out 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
    
      * 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