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Release notes for GStreamer Base Plugins 1.2.1

The GStreamer team is proud to announce a new bug-fix release in the 1.x stable series of the core of the GStreamer streaming media framework.

The 1.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.10.x series. It is, however, parallel installable with the 0.10.x series and will not affect an existing 0.10.x installation.

This module contains a set of reference plugins, base classes for other plugins, and helper libraries. It also includes essential elements such as audio and video format converters, and higher-level components like playbin, decodebin, encodebin, and discoverer. 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, giosrc
  • network: tcp
  • typefind functions
  • audio processing: audioconvert, adder, audiorate, audioresample, volume
  • visualisation: libvisual
  • video processing: videoconvert, videoscale
  • high-level components: playbin, uridecodebin, decodebin, encodebin, discoverer
  • libraries: app, audio, fft, pbutils, riff, rtp, rtsp, sdp, tag, video
Other modules containing plugins are:
contains a set of well-supported plugins under our preferred license
contains a set of well-supported plugins, but might pose problems for distributors
contains a set of less supported plugins that haven't passed the rigorous quality testing we expect, or are still missing documentation and/or unit tests
contains a set of codecs plugins based on libav (formerly gst-ffmpeg)

Bugs fixed in this release

  • 701813 : Reverse playback not working with videotestsrc
  • 708689 : rtspconnection: RTSP watch is dispatched after closing the connection
  • 708773 : pbutils: add MPEG 2 AAC description
  • 708789 : playbin: make sure elements are in null before disposing
  • 708880 : rtspconnection: Not connecting to proxy when specified
  • 708952 : audio: change buffer ts when clipping buffer even if data length is same
  • 708953 : audiorate: clip buffers before pushing them out
  • 708954 : pbutils: add entry for text/x-raw
  • 709210 : Hangs on startup getting PulseAudio volume
  • 709408 : audioconvert: modifies buffer mapped for READ
  • 709637 : oggmux: Make sure we end up sending EOS if we received EOS on all sinkpads
  • 709754 : audioringbuffer: Clears need_reorder flag wrongly
  • 709938 : navigation: Missing gobject-introspection annotations
  • 710325 : playback: Add subpicture/x-dvb as raw caps
  • 711003 : videoscale: borders are filled with green when using NV12 pixelformat
  • 711231 : rtspconnection: allow setting tls certificate validation flags
  • 711550 : appsrc: Deadlocking because holding mutex while setting caps


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

  • Aleix Conchillo Flaque
  • Antonio Ospite
  • Hans Månsson
  • Matej Knopp
  • Ognyan Tonchev
  • Sebastian Dröge
  • Stefan Sauer
  • Stephan Sundermann
  • Takashi Iwai
  • Thiago Santos
  • Thibault Saunier
  • Tim-Philipp Müller
  • Tom Greenwood

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