RELEASE: GStreamer Base Plug-ins 0.10.15 'Light Years Ahead'



This mail announces the release of GStreamer Base Plug-ins 0.10.15 'Light Years Ahead'.


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 0.10.15 "No need to argue"
        


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
    
      * 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
      * 

Bugs fixed in this release
     
      * 475395 : decodebin2 leaks request-pads
      * 475451 : [decodebin2] leaks ghostpad
      * 378770 : [xvimagesink] race condition in event thread?
      * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
      * 430677 : [audioconvert] does not preserve channel positions when f...
      * 442654 : [volume] controller bypassed by default
      * 445529 : [volume] support for 24/32-bit audio/x-raw-int
      * 446766 : return code for gst_base_rtp_payload_audio_handle_event()
      * 451970 : Subparse requires HTML parser
      * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
      * 459334 : [textoverlay] expose pango line alignment property
      * 459585 : [basertpdepayload] api without namespace
      * 460422 : [audiotestsrc] Add support for float and double output
      * 462805 : [alsa] compilation fails with gcc 4.2
      * 462979 : Add 'silent' property to GstTimeOverlay
      * 463215 : [audioconvert] compile errors
      * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
      * 464666 : [playbin] QT trailer hangs in preroll with decodebin2
      * 464690 : Add connection-speed property to uridecodebin element
      * 465015 : [playbin] Not removed probes causes deadlocks in streamin...
      * 465028 : some warnings with mingw
      * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
      * 468129 : [basertpaudiopayload] event handler returns the wrong value
      * 468619 : New library gstfft: FFT library for integer and float typ...
      * 470456 : [API] add gst_missing_*_installer_detail_new()
      * 470766 : [ssaparse] line breaks in SSA subtitle parser
      * 471067 : Make the SDP code useable for generating SDP descriptions
      * 471194 : [rtpbuffer] RTP headers are wrong for win32
      * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
      * 474384 : gstrtsp-enumtypes.c and .h needed for win32
      * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
      * 475731 : rtspconnection is able to read incomplete messages
      * 483620 : All Rtp buffers are discarded --  gst_rtp_buffer_get_payl...
      * 484989 : memleak, not unrefed caps for gstbasertppayload.c
      * 489010 : Please change default channel order for WAVE_EXT-less .wa...
      * 491722 : [playbin] regression: crash with external subtitles
      * 492098 : [GstFFT] Broken scaling
      * 492114 : Build issues on Windows/MSVC
      * 492306 : compilation errors with MinGW
      * 492813 : Missing symbols in libgstrtp.def
      * 493986 : Build issues on Windows (missing symbols)
      * 494346 : pre-release vs6 patch
      * 496548 : Including malloc.h breaks macos build
      * 496724 : DSW file references non-existent DSP files
      * 464079 : audiotestsrc doesn't respond to conversion queries properly
      * 442065 : floatcast.h includes config.h and might break other apps
      * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
      * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
      * 464028 : Move connection-speed from playbin to playbasebin

API changed in this release
     
- API additions:
    
* GstTagDemux base class for simple tag demuxers
* GstBaseAudioSrc::provide-clock property
* gst_rtcp_ntp_to_unix()
* gst_rtcp_unix_to_ntp()
* gst_rtp_buffer_get_header_len()
* gst_rtp_buffer_get_extension_data()
* gst_rtp_buffer_compare_seqnum()
* gst_rtp_buffer_ext_timestamp()
* gst_rtcp_packet_sdes_copy_entry()
* gst_install_plugins_supported()
* gst_missing_*_installer_detail_new() convenience API
* gst_rtsp_connection_poll()
* GstTextOverlay::line-alignment property

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
    
      * Stefan Kost
      * Alexander Shopov
      * Damien Lespiau
      * Dan Williams
      * Daniel Díaz
      * David Schleef
      * Davyd Madeley
      * Funda Wang
      * Haakon Sporsheim
      * Ilkka Tuohela
      * Jakub Bogusz
      * Jan Schmidt
      * Jason Kivlighn
      * Jens Granseuer
      * Johan Dahlin
      * Jorge González González
      * Josep Torra Valles
      * Julien MOUTTE
      * Laurent Glayal
      * Michael Smith
      * Mogens Jaeger
      * Ole André Vadla Ravnås
      * Olivier Crete
      * Peter Kjellerstedt
      * Renato Filho
      * René Stadler
      * Sebastian Dröge
      * Sebastien Moutte
      * Stefan Kost
      * Thijs Vermeir
      * Thomas Vander Stichele
      * Tim-Philipp Müller
      * Tommi Myöhänen
      * Vincent Torri
      * Wim Taymans
      * Yang Hong
 


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