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
This module contains elements for, among others:
Other modules containing plugins are:
- 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
- contains a set of well-supported plugins under our preferred license
- contains a set of well-supported plugins, but might pose problems for
- 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
: Reverse playback not working with videotestsrc
: rtspconnection: RTSP watch is dispatched after closing the connection
: pbutils: add MPEG 2 AAC description
: playbin: make sure elements are in null before disposing
: rtspconnection: Not connecting to proxy when specified
: audio: change buffer ts when clipping buffer even if data length is same
: audiorate: clip buffers before pushing them out
: pbutils: add entry for text/x-raw
: Hangs on startup getting PulseAudio volume
: audioconvert: modifies buffer mapped for READ
: oggmux: Make sure we end up sending EOS if we received EOS on all sinkpads
: audioringbuffer: Clears need_reorder flag wrongly
: navigation: Missing gobject-introspection annotations
: playback: Add subpicture/x-dvb as raw caps
: videoscale: borders are filled with green when using NV12 pixelformat
: rtspconnection: allow setting tls certificate validation flags
: 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 http://gstreamer.freedesktop.org.
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).
There is also a #gstreamer IRC channel on the Freenode IRC network.
Git is hosted on git.freedesktop.org. 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