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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 gst-rtsp-server download directory.

The git repository and details how to clone it can be found at git.freedesktop.org .

Homepage

The project's website is https://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.

Developers

Git is hosted on git.freedesktop.org. You can browse the gst-rtsp-server 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.

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

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