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4.2 KiB
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115 lines
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4.2 KiB
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Release notes for GStreamer RTSP Server Library 1.5.1
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The GStreamer team is pleased to announce the first release of the unstable
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1.5 release series. The 1.5 release series is adding new features on top of
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the 1.0, 1.2 and 1.4 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 1.5 release series
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will lead to the stable 1.6 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 1.5 release series.
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Features of this release
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Bugs fixed in this release
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* 732238 : Listen on the multicast group for RTP/RTCP packets
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* 734546 : tests: Unref element after usage
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* 736041 : Protect rtsp transport data.
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* 736647 : Tunneled RTSP sessions do not always timeout as expected
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* 737110 : rtsp-client: race condition when closing client connection
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* 737631 : gst-rtsp-server deadlock while sending response over TCP
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* 737675 : media: media_unprepare() is kind of broken
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* 737690 : rtsp-client: deadlock when setting session medias to NULL
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* 737797 : rtsp-stream: lock not released when leaving bin and transports not removed
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* 737829 : rtsp-server: deactivate media when shutting down from paused
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* 738905 : rtsp-client: add stream transport to the context
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* 739112 : rtsp-client: Can not allocate ports for interleaved traffic in setup
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* 740752 : add retransmission support
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* 740845 : crash when reciving a rtcp after teardown but before client finalize.
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* 741678 : configure: add --disable-examples switch
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* 742115 : Examples: Accept a 'port' argument for running multiple instances
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* 742869 : Remove URI-escaping of RTSP session-id
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* 742954 : Crash when two treads are in handle_new_sample at the same time.
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* 743175 : Add support for RECORD
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* 743346 : When system time is increased the ongoing RTSP sessions will time out.
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* 743734 : RTCP packets not sent
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* 744379 : gst-rtsp-server does not preroll when piping data into the media-pipeline
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* 745704 : Losing the first packet
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* 747614 : gst-rtsp-server: uninitialized clock rate causes critical warning
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* 747839 : gst-rtsp-server: doesn't perform retransmission to both streams in test-video-rtx
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* 748058 : autogen.sh fails due to autopoint erroring out due to missing gettext version in configure.ac
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* 749845 : Client have problem to find the teardown 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 Flaqué
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* Alistair Buxton
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* Andreas Frisch
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* Anila Balavan
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* Arun Raghavan
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* Branko Subasic
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* Edward Hervey
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* Gregor Boirie
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* Göran Jönsson
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* Hyunjun Ko
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* Jan Schmidt
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* Kent-Inge Ingesson
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* Linus Svensson
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* Luis de Bethencourt
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* Matthew Waters
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* Nicolas Dufresne
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* Nirbheek Chauhan
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* Ognyan Tonchev
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* Olivier Crête
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* Sebastian Dröge
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* Sebastian Rasmussen
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* Srimanta Panda
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* Stefan Sauer
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* Tim-Philipp Müller
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* Vincent Penquerc'h
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* Wim Taymans
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