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Release notes for GStreamer Base Plugins 1.3.3
The GStreamer team is pleased to announce the third release of the unstable
1.3 release series. The 1.3 release series is adding new features on top of
the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release
series of the GStreamer multimedia framework. The unstable 1.3 release series
will lead to the stable 1.4 release series in the next weeks, and newly added
API can still change until that point.
This is hopefully the last 1.3 development release and will be followed by
the first 1.4.0 release candidate (1.3.90) in 1-2 weeks. Which then hopefully
is followed by 1.4.0 soonish in early July.
Binaries for Android, iOS, Mac OS X and Windows will be provided separately
during the unstable 1.3 release series.
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:
gst-plugins-good
contains a set of well-supported plugins under our preferred license
gst-plugins-ugly
contains a set of well-supported plugins, but might pose problems for
distributors
gst-plugins-bad
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
gst-libav
contains a set of codecs plugins based on libav (formerly gst-ffmpeg)
Bugs fixed in this release
* 709868 : Keep still meaningfull pending events on FLUSH_STOP
* 724231 : appsrc: handle flushing from send_event
* 730559 : dmabuf: fix checking mmap flags
* 730749 : Failed to determine keyframeness of audio/x-opus packet
* 730868 : uridecodebin: Does not handle RTSP streams where one of the payload formats is not supported properly
* 730874 : audio: Add a missing precondition to gst_audio_format_from_string()
* 731121 : alsasink: Race condition causes alsasink to use invalid caps when a pipeline fails to start
* 731566 : tcpserversrc: close the server socket after accepting a connection
* 731567 : tcpserversrc: return GST_FLOW_FLUSHING instead of GST_FLOW_ERROR when accept is canceled
==== Download ====
You can find source releases of gst-plugins-base in the download
directory: http://gstreamer.freedesktop.org/src/gst-plugins-base/
The git repository and details how to clone it can be found at
http://cgit.freedesktop.org/gstreamer/gst-plugins-base/
==== Homepage ====
The project's website is 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
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.
==== Developers ====
GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned
from there (see link above).
Interested developers of the core library, plugins, and applications should
subscribe to the gstreamer-devel list.
Contributors to this release
* Edward Hervey
* Michael Olbrich
* Philip Withnall
* Sebastian Dröge
* Thiago Santos
* Thibault Saunier
* Tim-Philipp Müller
* Vincent Penquerc'h
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