gstreamer/RELEASE
Sebastian Dröge 525339790c Release 1.3.1
2014-05-03 18:40:24 +02:00

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Release notes for GStreamer RTSP Server Library 1.3.1
The GStreamer team is pleased to announce the first 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.
Binaries for Android, iOS, Mac OS X and Windows will be provided separately
during the unstable 1.3 release series.
The versioning scheme that is used in general is that 1.x.y is API and
ABI backwards compatible with previous 1.x.y releases. If x is an even
number it is a stable release series and all releases in this series
will only contain important bugfixes, e.g. the 1.0 series with 1.0.7. If
x is odd it is a development release series that will lead to the next
stable release series 1.x+1 and contains new features and bigger
changes. During the development release series, new API can still
change.
Bugs fixed in this release
* 725484 : gst-rtsp-server: Ignore gcov intermediate files
* 725528 : rtspserver: Enable and fix gtk-doc warnings
* 725879 : rtsp-client: headers in GET response not configurable for tunnels
* 726362 : rtsp-stream: fix a typo where IPv4 and IPv6 addresses were confused.
* 726470 : tests: Add unit tests for sessionpool
* 726873 : rtsp-threadpool: Improve code coverage of check tests
* 726940 : rtsp-session-media: add more tests to improve code coverage
* 726941 : docs: Add annotations to support language bindings
* 727102 : rtsp-media: deadlock with dynamic pipelines when preroll fails
* 727231 : rtsp-server: The media streams leak
* 727376 : crash if media_prepare() fails to allocate UDP ports
* 727488 : There is a race when disconnecting POST channel in tunneled mode
* 728029 : rtsp-media: Make media_prepare() virtual
* 728060 : rtsp-session-pool: Incorrect annotation and leak in unit test
* 728153 : Problem with send_lock when data in backlog and recive a teardown request.
* 728970 : rtsp-client: add signal before sending response
==== Download ====
You can find source releases of gst-rtsp-server in the download
directory: http://gstreamer.freedesktop.org/src/gst-rtsp-server/
The git repository and details how to clone it can be found at
http://cgit.freedesktop.org/gstreamer/gst-rtsp-server/
==== 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.
Applications
Contributors to this release
* Aleix Conchillo Flaque
* Aleix Conchillo Flaqué
* Alessandro Decina
* Alexander Schrab
* Andrey Utkin
* Branko Subasic
* David Schleef
* David Svensson Fors
* Edward Hervey
* Emmanuel Pacaud
* Fabian Deutsch
* George McCollister
* Göran Jönsson
* Jonas Holmberg
* Linus Svensson
* Lubosz Sarnecki
* Luis de Bethencourt
* Mark Nauwelaerts
* Miguel Angel Cabrera Moya
* Ognyan Tonchev
* Olivier Crête
* Patricia Muscalu
* Patrick Radizi
* Robert Krakora
* Sebastian Dröge
* Sebastian Pölsterl
* Sebastian Rasmussen
* Stefan Kost
* Stefan Sauer
* Thijs Vermeir
* Thomas Vander Stichele
* Tim-Philipp Müller
* Victor Gottardi
* Vincent Penquerc'h
* Wim Taymans
* Youness Alaoui
* mat