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Release 1.3.2
This commit is contained in:
Sebastian Dröge
parent
afa3f3f48a
commit
acbdc49c35
6 changed files with 123 additions and 64 deletions
83
ChangeLog
83
ChangeLog
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@ -1,9 +1,86 @@
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=== release 1.3.1 ===
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=== release 1.3.2 ===
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2014-05-03 Sebastian Dröge <slomo@coaxion.net>
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2014-05-21 Sebastian Dröge <slomo@coaxion.net>
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* configure.ac:
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releasing 1.3.1
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releasing 1.3.2
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2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
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* common:
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Automatic update of common submodule
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From 211fa5f to 1f5d3c3
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2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
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* gst/rtsp-server/rtsp-client.c:
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client: store TCP ports in transport
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Store the TCP ports in the transport when we are doing RTSP over TCP.
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This way, we can easily get to the ports from the transport.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
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2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: add signals for new RTP/RTCP encoders
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New signals to allow the user to configure the dynamically created
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encoders.
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https://bugzilla.gnome.org/show_bug.cgi?id=730228
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2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-media.h:
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media: Make suspend()/unsuspend() virtual
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
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2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
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* gst/rtsp-server/rtsp-client.c:
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client: fix send-message signal marshaller
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Use generic marshalling for the send-message signal. It has
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two POINTER arguments, not just one.
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https://bugzilla.gnome.org/show_bug.cgi?id=729900
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2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
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* tests/check/gst/media.c:
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tests: add and remove pads only once
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In this test we simulate a dynamic pad by watching the caps event.
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Because of renegotiation in the base payloader now, this caps is sent
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multiple times but we can only deal with 1 invocation, use a variable to
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only 'add and remove' the pad once.
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2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
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* tests/check/gst/rtspserver.c:
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tests: add unit test for correct handling of Require headers
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https://bugzilla.gnome.org/show_bug.cgi?id=729426
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2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
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Servers must handle Require headers and must report a failure
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if they don't handle any of the Required options, see RFC 2326,
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section 12.32: https://tools.ietf.org/html/rfc2326#page-54
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https://bugzilla.gnome.org/show_bug.cgi?id=729426
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2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
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Back to development
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=== release 1.3.1 ===
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2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
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* ChangeLog:
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* NEWS:
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* RELEASE:
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* configure.ac:
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* gst-rtsp-server.doap:
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Release 1.3.1
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2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
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18
NEWS
18
NEWS
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@ -1,4 +1,4 @@
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This is GStreamer RTSP Server 1.3.1
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This is GStreamer RTSP Server 1.3.2
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Changes since 1.2:
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@ -45,6 +45,8 @@ New API:
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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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• GstDiscoverer has new and simplified API to get details about missing
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plugins and information to pass to the plugin installer.
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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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@ -62,6 +64,14 @@ Major changes:
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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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∘ New downloadbuffer element that replaces the download
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buffering feature of queue2. Compared to queue2's code
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it is much simpler and only for this single use case.
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A noteworthy new feature is that it's downloading gaps
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in the already downloaded stream parts when nothing else
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is to be downloaded.
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This is now used by playbin when download buffering is
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enabled.
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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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@ -78,7 +88,7 @@ Major changes:
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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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∘ gst-libav now uses libav 10.1, 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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@ -95,6 +105,9 @@ Major changes:
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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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∘ dashdemux and mssdemux are now also pushing fragments downstream
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while they're downloaded instead of waiting for each fragment to
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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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@ -107,4 +120,3 @@ Things to look out for:
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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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60
RELEASE
60
RELEASE
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@ -1,8 +1,8 @@
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Release notes for GStreamer RTSP Server Library 1.3.1
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Release notes for GStreamer RTSP Server Library 1.3.2
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The GStreamer team is pleased to announce the first release of the unstable
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The GStreamer team is pleased to announce the second 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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@ -28,24 +28,16 @@ change.
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Features of this release
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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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* 729426 : Should respond " 551 Option not supported " in case a Require header is received
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* 729776 : Set client port from URL
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* 729900 : rtsp-client: wrong marshalling in send-message signal
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* 730109 : media: Make suspend()/unsuspend() virtual
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* 730228 : stream: add signals for new RTP/RTCP encoders
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==== Download ====
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@ -84,41 +76,9 @@ 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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2
common
2
common
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@ -1 +1 @@
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Subproject commit 1f5d3c3163cc3399251827235355087c2affa790
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Subproject commit 211fa5f2d0930dfd6891b386d42edba6d88c2a19
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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], [1.3.1.1],
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AC_INIT([GStreamer RTSP Server Library], [1.3.2],
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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, 301, 0, 301)
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AS_LIBTOOL(GST, 302, 0, 302)
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dnl *** required versions of GStreamer stuff ***
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GST_REQ=1.3.1.1
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GSTPB_REQ=1.3.1.1
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GSTPG_REQ=1.3.1.1
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GSTPD_REQ=1.3.1.1
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GST_REQ=1.3.2
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GSTPB_REQ=1.3.2
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GSTPG_REQ=1.3.2
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GSTPD_REQ=1.3.2
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dnl *** autotools stuff ****
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@ -28,7 +28,17 @@ RTSP server library based on GStreamer
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<location rdf:resource="git://anongit.freedesktop.org/gstreamer/gst-rtsp-server"/>
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<browse rdf:resource="http://cgit.freedesktop.org/gstreamer/gst-rtsp-server"/>
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</GitRepository>
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</repository>
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</repository>
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<release>
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<Version>
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<revision>1.3.2</revision>
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<branch>1.3</branch>
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<name></name>
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<created>2014-05-21</created>
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<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.2.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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