2014-05-03 16:40:24 +00:00
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Release notes for GStreamer RTSP Server Library 1.3.1
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The GStreamer team is pleased to announce the first release of the unstable
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1.3 release series. The 1.3 release series is adding new features on top of
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the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release
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series of the GStreamer multimedia framework. The unstable 1.3 release series
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will lead to the stable 1.4 release series in the next weeks, and newly added
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API can still change until that point.
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Binaries for Android, iOS, Mac OS X and Windows will be provided separately
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during the unstable 1.3 release series.
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The versioning scheme that is used in general is that 1.x.y is API and
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ABI backwards compatible with previous 1.x.y releases. If x is an even
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number it is a stable release series and all releases in this series
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will only contain important bugfixes, e.g. the 1.0 series with 1.0.7. If
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x is odd it is a development release series that will lead to the next
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stable release series 1.x+1 and contains new features and bigger
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changes. During the development release series, new API can still
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change.
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Bugs fixed in this release
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* 725484 : gst-rtsp-server: Ignore gcov intermediate files
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* 725528 : rtspserver: Enable and fix gtk-doc warnings
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* 725879 : rtsp-client: headers in GET response not configurable for tunnels
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* 726362 : rtsp-stream: fix a typo where IPv4 and IPv6 addresses were confused.
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* 726470 : tests: Add unit tests for sessionpool
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* 726873 : rtsp-threadpool: Improve code coverage of check tests
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* 726940 : rtsp-session-media: add more tests to improve code coverage
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* 726941 : docs: Add annotations to support language bindings
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* 727102 : rtsp-media: deadlock with dynamic pipelines when preroll fails
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* 727231 : rtsp-server: The media streams leak
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* 727376 : crash if media_prepare() fails to allocate UDP ports
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* 727488 : There is a race when disconnecting POST channel in tunneled mode
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* 728029 : rtsp-media: Make media_prepare() virtual
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* 728060 : rtsp-session-pool: Incorrect annotation and leak in unit test
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* 728153 : Problem with send_lock when data in backlog and recive a teardown request.
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* 728970 : rtsp-client: add signal before sending response
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==== Download ====
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You can find source releases of gst-rtsp-server in the download
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directory: http://gstreamer.freedesktop.org/src/gst-rtsp-server/
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The git repository and details how to clone it can be found at
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http://cgit.freedesktop.org/gstreamer/gst-rtsp-server/
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==== Homepage ====
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The project's website is http://gstreamer.freedesktop.org/
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==== Support and Bugs ====
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We use GNOME's bugzilla for bug reports and feature requests:
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http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer
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Please submit patches via bugzilla as well.
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For help and support, please subscribe to and send questions to the
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gstreamer-devel mailing list (see below for details).
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There is also a #gstreamer IRC channel on the Freenode IRC network.
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==== Developers ====
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GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned
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from there (see link above).
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Interested developers of the core library, plugins, and applications should
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subscribe to the gstreamer-devel list.
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Applications
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Contributors to this release
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* Aleix Conchillo Flaque
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* Aleix Conchillo Flaqué
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* Alessandro Decina
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* Alexander Schrab
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* Andrey Utkin
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* Branko Subasic
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* David Schleef
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* David Svensson Fors
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* Edward Hervey
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* Emmanuel Pacaud
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* Fabian Deutsch
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* George McCollister
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* Göran Jönsson
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* Jonas Holmberg
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* Linus Svensson
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* Lubosz Sarnecki
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* Luis de Bethencourt
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* Mark Nauwelaerts
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* Miguel Angel Cabrera Moya
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* Ognyan Tonchev
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* Olivier Crête
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* Patricia Muscalu
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* Patrick Radizi
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* Robert Krakora
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* Sebastian Dröge
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* Sebastian Pölsterl
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* Sebastian Rasmussen
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* Stefan Kost
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* Stefan Sauer
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* Thijs Vermeir
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* Thomas Vander Stichele
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* Tim-Philipp Müller
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* Victor Gottardi
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* Vincent Penquerc'h
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* Wim Taymans
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* Youness Alaoui
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* mat
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