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Release 1.3.1
This commit is contained in:
Sebastian Dröge
parent
dc27833c1f
commit
525339790c
5 changed files with 7501 additions and 7 deletions
111
NEWS
111
NEWS
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@ -1 +1,110 @@
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This is GstRTSP
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This is GStreamer RTSP Server 1.3.1
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Changes since 1.2:
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New API:
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• GstMessageType has GST_MESSAGE_EXTENDED added. All types before
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that can be used together as a flags type as before, but from
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that message onwards the types are just counted incrementally.
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This was necessary to be able to add more message types.
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In 2.0 GstMessageType will just become an enum and not a flags
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type anymore.
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• GstDeviceMonitor for device probing, e.g. to list all available
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audio or video capture devices. This is the replacement for
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GstPropertyProbe from 0.10.
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• Events accumulate the running-time offset now when travelling
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through pads, as set by the gst_pad_set_offset() function. This
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allows to compensate for this in the QOS event for example.
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• GstBuffer has a new flag "tag-memory" that is set automatically
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when memory is added or removed to a buffer. This allows buffer
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pools to detect if they can recycle a buffer or need to reset
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it first.
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• GstToc has new API to mark GstTocEntries as loops.
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• A not-authorized resource error has been defined to notify
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applications that accessing the resource has failed because
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of missing authorization and to distinguish this case from others.
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This change is actually already in 1.2.4.
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• GstPad has a new flag "accept-intersect", that will let the default
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ACCEPT_CAPS query handler do an intersection instead of subset check.
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This is interesting for parser elements that can handle incomplete
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caps.
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• GstCollectPads has support for flushing and a default handler for
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SEEK events now.
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• GstSegment has new API to offset the running time by a specific
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value and this is used in GstPad to allow positive and negative
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offsets in gst_pad_set_offset() in all situations.
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• Support for h265/HEVC and VP8 has been added to the codec utils and codec
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parsers library, and was integrated into various elements.
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• API for adjusting the TLS validation of RTSP connection has been added.
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• The RTSP and SDP library has MIKEY (RFC 3830) support now, and
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there is API to distinguish between the different RTSP profiles.
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• API to access RTP time information and statistics.
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• Support for auxiliary streams was added to rtpbin.
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• Support for tiled, raw video formats has been added.
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• GstVideoDecoder and GstAudioDecoder have API to help aggregating tag
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events and merge custom tags into them consistently.
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• playbin/playsink has support for application provided audio and video
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filters.
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• The GL library was merged from gst-plugins-gl to gst-plugins-bad,
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providing a generic infrastructure for handling GL inside GStreamer
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pipelines and a plugin with some elements using these, especially
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a video sink. Supported platforms currently are Android, Cocoa (OS X),
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DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11,
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Wayland and EGL platforms.
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This replaces eglglessink and also is supposed to replace osxvideosink.
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Major changes:
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• New plugins and elements:
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∘ v4l2videodec element for accessing hardware codecs on
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platforms that make them accessible via V4L2, e.g.
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Samsung Exynos. This comes together with major refactoring
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of the existing V4L2 elements and the corresponding
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infrastructure.
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The v4l2videodec element replaces the mfcdec element.
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∘ rtpstreampay and rtpstreamdepay elements for transmitting
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RTP packets over a stream API (e.g. TCP) according to
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RFC 4571.
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∘ rtprtx elements for standard compliant implementation of
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retransmissions, integrated into the rtpmanager plugin.
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∘ audiomixer element that mixes multiple audio streams together
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into a single one while keeping synchronization. This is
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planned to become the replacement of the adder element.
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∘ OpenNI2 plugin for 3D cameras like the Kinect camera.
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∘ OpenEXR plugin for decoding high-dynamic-range EXR images.
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∘ curlsshsink and curlsftpsink to write files via SSH/SFTP.
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∘ videosignal, ivfparse and sndfile plugins ported from 0.10.
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∘ avfvideosrc, vtdec and other elements were ported from 0.10 and
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are available on OS X and iOS now.
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• Other changes:
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∘ gst-libav now uses libav 10, and gained support for H265/HEVC.
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∘ Support for hardware codecs and special memory types has been
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improved with bugfixes and feature additions in various plugins
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and base classes.
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∘ Various bugfixes and improvements to buffering in queue2 and
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multiqueue elements.
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∘ dvbsrc supports more delivery mechanisms and other features
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now, including DVB S2 and T2 support.
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∘ The MPEGTS library has support for many more descriptors.
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∘ Major improvements to tsdemux, especially time related.
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∘ souphttpsrc now has support for keep-alive connections,
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compression, configurable number of retries and configuration
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for SSL certificate validation.
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∘ hlsdemux has undergone major refactoring and works more
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reliable now and supports more HLS features like trick modes.
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Also fragments are pushed downstream while they're downloaded
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now instead of waiting for each fragment to finish.
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∘ videoflip can automatically flip based on the orientation tag.
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∘ openjpeg supports the OpenJPEG2 API.
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∘ gst-rtsp-server supports SRTP and MIKEY now.
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∘ Lots of fixes for coverity warnings all over the place.
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∘ 400+ fixed bug reports, and many other bug fixes and other
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improvements everywhere that had no bug report.
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Things to look out for:
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• The eglglessink element was removed and replaced by the glimagesink
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element.
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• The mfcdec element was removed and replaced by v4l2videodec.
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• osxvideosink is only available in OS X 10.6 or newer.
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124
RELEASE
124
RELEASE
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@ -0,0 +1,124 @@
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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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12
configure.ac
12
configure.ac
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@ -2,7 +2,7 @@ AC_PREREQ(2.62)
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dnl initialize autoconf
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dnl when going to/from release please set the nano (fourth number) right !
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dnl releases only do Wall, cvs and prerelease does Werror too
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AC_INIT([GStreamer RTSP Server Library], [1.3.0.1],
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AC_INIT([GStreamer RTSP Server Library], [1.3.1],
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[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
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[gst-rtsp-server])
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AG_GST_INIT
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@ -53,13 +53,13 @@ dnl 1.2.5 => 205
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dnl 1.10.9 (who knows) => 1009
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dnl
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dnl sets GST_LT_LDFLAGS
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AS_LIBTOOL(GST, 300, 0, 300)
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AS_LIBTOOL(GST, 301, 0, 301)
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dnl *** required versions of GStreamer stuff ***
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GST_REQ=1.3.0.1
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GSTPB_REQ=1.3.0.1
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GSTPG_REQ=1.3.0.1
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GSTPD_REQ=1.3.0.1
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GST_REQ=1.3.1
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GSTPB_REQ=1.3.1
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GSTPG_REQ=1.3.1
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GSTPD_REQ=1.3.1
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dnl *** autotools stuff ****
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@ -30,6 +30,16 @@ RTSP server library based on GStreamer
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</GitRepository>
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</repository>
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<release>
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<Version>
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<revision>1.3.1</revision>
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<branch>1.3</branch>
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<name></name>
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<created>2014-05-03</created>
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<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.1.tar.xz" />
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</Version>
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</release>
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<release>
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<Version>
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<revision>1.1.90</revision>
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