Release 1.3.1

NEWS

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

RELEASE

@@ -0,0 +1,124 @@
+
+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
+ 

configure.ac

@@ -2,7 +2,7 @@ AC_PREREQ(2.62)
 dnl initialize autoconf
 dnl when going to/from release please set the nano (fourth number) right !
 dnl releases only do Wall, cvs and prerelease does Werror too
-AC_INIT([GStreamer RTSP Server Library], [1.3.0.1],
+AC_INIT([GStreamer RTSP Server Library], [1.3.1],
     [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
     [gst-rtsp-server])
 AG_GST_INIT
@@ -53,13 +53,13 @@ dnl      1.2.5 => 205
 dnl      1.10.9 (who knows) => 1009
 dnl
 dnl sets GST_LT_LDFLAGS
-AS_LIBTOOL(GST, 300, 0, 300)
+AS_LIBTOOL(GST, 301, 0, 301)
 
 dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.3.0.1
+GST_REQ=1.3.1
-GSTPB_REQ=1.3.0.1
+GSTPB_REQ=1.3.1
-GSTPG_REQ=1.3.0.1
+GSTPG_REQ=1.3.1
-GSTPD_REQ=1.3.0.1
+GSTPD_REQ=1.3.1
 
 dnl *** autotools stuff ****
 

gst-rtsp-server.doap

@@ -30,6 +30,16 @@ RTSP server library based on GStreamer
    </GitRepository>
  </repository> 
 
+ <release>
+  <Version>
+   <revision>1.3.1</revision>
+   <branch>1.3</branch>
+   <name></name>
+   <created>2014-05-03</created>
+   <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.1.tar.xz" />
+  </Version>
+ </release>
+
  <release>
   <Version>
    <revision>1.1.90</revision>