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Release notes for GStreamer Base Plug-ins 0.10.13 "What's Going on?"

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:
contains a set of well-supported plug-ins under our preferred license
contains a set of well-supported plug-ins, but might pose problems for distributors
contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect

Features of this release

  • Many fixes and improvements
  • RTP and RTCP support improved
  • Parallel installability with 0.8.x series
  • Threadsafe design and API

Bugs fixed in this release

  • 339838 : [audioconvert] support floats with non-native endianness
  • 393975 : closing x/xvimagesink window crashes gst-launch
  • 405072 : [API] add gst_tag_freeform_string_to_utf8()
  • 413799 : [subparse] add support for MPL2 format
  • 414645 : GstMixerTrack should make untranslated label available
  • 420079 : [audioconvert] Uses biased rounding which results in dist...
  • 420578 : [subparse] add more colour map in sami parser
  • 421834 : videorate breaks on dimension changes
  • 423051 : Vorbis tags of type double use locale-dependent formatting
  • 423055 : Verify ReplayGain vorbistag processing in libs/tag testsuite
  • 425455 : Decodebin2 leaks pads
  • 426250 : GstPlayBaseBin leaks streaminfo objects
  • 428187 : Rtp base depayloader class doesn't send new_segment after...
  • 431672 : gst_base_rtp_audio_payload_push() should take object of i...
  • 432362 : [ximagesink] doesn't build if XShm is not available
  • 432755 : [videorate] leaks buffer if flow != OK
  • 432984 : [baseaudiosrc] misleading warning message when dropping s...
  • 433888 : [theoradec] does not generate a perfect stream
  • 436562 : Theoradec doesn't work well with gnonlin
  • 438840 : [theoradec] does not compile with old version of libtheora
  • 440997 : [gstriff] Doesn't handle width!=depth files with audio/x-...
  • 441295 : audioconvert doesn't build on VS6
  • 442024 : regression in playbin buffering
  • 350299 : [playbin] " Internal data flow error " opening movie with s...
  • 410039 : totem crashed with SIGSEGV in new_decoded_pad_full()
  • 340842 : do latency calculation for live sources
  • 341078 : RB does not play beyond initially downloaded podcast file
  • 414496 : [id3demux, id3v2mux] Add support for GST_TAG_MUSICBRAINZ_...

API changes

  • API additions
    • add gst_tag_freeform_string_to_utf8()
    • GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
    • GstBaseAudioSink::slave-method property
    • add "min-ptime" property to RTP base audio payloader
    • gst_base_rtp_audio_payload_push()
    • gst_base_rtp_audio_payload_get_adapter()
    • GstMixerTrack::untranslated-label property


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 .


The project's website is

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.


Git is hosted on 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.


Contributors to this release

  • Alex Lancaster
  • Andy Wingo
  • Christian Kirbach
  • Dan Williams
  • David Schleef
  • Edward Hervey
  • Jan Schmidt
  • Julien MOUTTE
  • Kamil Pawlowski
  • Marc-Andre Lureau
  • Mark Nauwelaerts
  • Michael Smith
  • Olivier Crete
  • René Stadler
  • Sebastian Dröge
  • Sebastien Moutte
  • Stefan Kost
  • Thomas Vander Stichele
  • Tim-Philipp Müller
  • Tommi Myöhänen
  • Vincent Torri
  • Wim Taymans
  • Young-Ho Cha
  • Zaheer Abbas Merali
  • Zeeshan Ali

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