mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-02-16 19:25:18 +00:00
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
|
|
@ -1,9 +1,86 @@
|
|||
=== release 1.3.1 ===
|
||||
=== release 1.3.2 ===
|
||||
|
||||
2014-05-03 Sebastian Dröge <slomo@coaxion.net>
|
||||
2014-05-21 Sebastian Dröge <slomo@coaxion.net>
|
||||
|
||||
* configure.ac:
|
||||
releasing 1.3.1
|
||||
releasing 1.3.2
|
||||
|
||||
2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* common:
|
||||
Automatic update of common submodule
|
||||
From 211fa5f to 1f5d3c3
|
||||
|
||||
2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* gst/rtsp-server/rtsp-client.c:
|
||||
client: store TCP ports in transport
|
||||
Store the TCP ports in the transport when we are doing RTSP over TCP.
|
||||
This way, we can easily get to the ports from the transport.
|
||||
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776
|
||||
|
||||
2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
|
||||
|
||||
* gst/rtsp-server/rtsp-stream.c:
|
||||
stream: add signals for new RTP/RTCP encoders
|
||||
New signals to allow the user to configure the dynamically created
|
||||
encoders.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=730228
|
||||
|
||||
2014-05-14 09:31:31 +0200 Ognyan Tonchev <ognyan@axis.com>
|
||||
|
||||
* gst/rtsp-server/rtsp-media.c:
|
||||
* gst/rtsp-server/rtsp-media.h:
|
||||
media: Make suspend()/unsuspend() virtual
|
||||
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
|
||||
|
||||
2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
|
||||
|
||||
* gst/rtsp-server/rtsp-client.c:
|
||||
client: fix send-message signal marshaller
|
||||
Use generic marshalling for the send-message signal. It has
|
||||
two POINTER arguments, not just one.
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=729900
|
||||
|
||||
2014-05-09 15:08:48 +0200 Wim Taymans <wtaymans@redhat.com>
|
||||
|
||||
* tests/check/gst/media.c:
|
||||
tests: add and remove pads only once
|
||||
In this test we simulate a dynamic pad by watching the caps event.
|
||||
Because of renegotiation in the base payloader now, this caps is sent
|
||||
multiple times but we can only deal with 1 invocation, use a variable to
|
||||
only 'add and remove' the pad once.
|
||||
|
||||
2014-05-02 20:06:29 +0100 Tim-Philipp Müller <tim@centricular.com>
|
||||
|
||||
* tests/check/gst/rtspserver.c:
|
||||
tests: add unit test for correct handling of Require headers
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=729426
|
||||
|
||||
2014-05-02 19:59:23 +0100 Tim-Philipp Müller <tim@centricular.com>
|
||||
|
||||
* gst/rtsp-server/rtsp-client.c:
|
||||
rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED
|
||||
Servers must handle Require headers and must report a failure
|
||||
if they don't handle any of the Required options, see RFC 2326,
|
||||
section 12.32: https://tools.ietf.org/html/rfc2326#page-54
|
||||
https://bugzilla.gnome.org/show_bug.cgi?id=729426
|
||||
|
||||
2014-05-03 20:48:43 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* configure.ac:
|
||||
Back to development
|
||||
|
||||
=== release 1.3.1 ===
|
||||
|
||||
2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
* ChangeLog:
|
||||
* NEWS:
|
||||
* RELEASE:
|
||||
* configure.ac:
|
||||
* gst-rtsp-server.doap:
|
||||
Release 1.3.1
|
||||
|
||||
2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
|
||||
|
||||
|
|
|
|||
18
NEWS
18
NEWS
|
|
@ -1,4 +1,4 @@
|
|||
This is GStreamer RTSP Server 1.3.1
|
||||
This is GStreamer RTSP Server 1.3.2
|
||||
|
||||
Changes since 1.2:
|
||||
|
||||
|
|
@ -45,6 +45,8 @@ New API:
|
|||
events and merge custom tags into them consistently.
|
||||
• playbin/playsink has support for application provided audio and video
|
||||
filters.
|
||||
• GstDiscoverer has new and simplified API to get details about missing
|
||||
plugins and information to pass to the plugin installer.
|
||||
• 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
|
||||
|
|
@ -62,6 +64,14 @@ Major changes:
|
|||
of the existing V4L2 elements and the corresponding
|
||||
infrastructure.
|
||||
The v4l2videodec element replaces the mfcdec element.
|
||||
∘ New downloadbuffer element that replaces the download
|
||||
buffering feature of queue2. Compared to queue2's code
|
||||
it is much simpler and only for this single use case.
|
||||
A noteworthy new feature is that it's downloading gaps
|
||||
in the already downloaded stream parts when nothing else
|
||||
is to be downloaded.
|
||||
This is now used by playbin when download buffering is
|
||||
enabled.
|
||||
∘ rtpstreampay and rtpstreamdepay elements for transmitting
|
||||
RTP packets over a stream API (e.g. TCP) according to
|
||||
RFC 4571.
|
||||
|
|
@ -78,7 +88,7 @@ Major changes:
|
|||
are available on OS X and iOS now.
|
||||
|
||||
• Other changes:
|
||||
∘ gst-libav now uses libav 10, and gained support for H265/HEVC.
|
||||
∘ gst-libav now uses libav 10.1, 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.
|
||||
|
|
@ -95,6 +105,9 @@ Major changes:
|
|||
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.
|
||||
∘ dashdemux and mssdemux are now also pushing fragments downstream
|
||||
while they're downloaded 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.
|
||||
|
|
@ -107,4 +120,3 @@ Things to look out for:
|
|||
element.
|
||||
• The mfcdec element was removed and replaced by v4l2videodec.
|
||||
• osxvideosink is only available in OS X 10.6 or newer.
|
||||
|
||||
|
|
|
|||
60
RELEASE
60
RELEASE
|
|
@ -1,8 +1,8 @@
|
|||
|
||||
Release notes for GStreamer RTSP Server Library 1.3.1
|
||||
Release notes for GStreamer RTSP Server Library 1.3.2
|
||||
|
||||
|
||||
The GStreamer team is pleased to announce the first release of the unstable
|
||||
The GStreamer team is pleased to announce the second 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
|
||||
|
|
@ -28,24 +28,16 @@ change.
|
|||
|
||||
|
||||
|
||||
Features of this release
|
||||
|
||||
|
||||
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
|
||||
* 729426 : Should respond " 551 Option not supported " in case a Require header is received
|
||||
* 729776 : Set client port from URL
|
||||
* 729900 : rtsp-client: wrong marshalling in send-message signal
|
||||
* 730109 : media: Make suspend()/unsuspend() virtual
|
||||
* 730228 : stream: add signals for new RTP/RTCP encoders
|
||||
|
||||
==== Download ====
|
||||
|
||||
|
|
@ -84,41 +76,9 @@ 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
|
||||
|
||||
2
common
2
common
|
|
@ -1 +1 @@
|
|||
Subproject commit 1f5d3c3163cc3399251827235355087c2affa790
|
||||
Subproject commit 211fa5f2d0930dfd6891b386d42edba6d88c2a19
|
||||
12
configure.ac
12
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], [1.3.1.1],
|
||||
AC_INIT([GStreamer RTSP Server Library], [1.3.2],
|
||||
[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, 301, 0, 301)
|
||||
AS_LIBTOOL(GST, 302, 0, 302)
|
||||
|
||||
dnl *** required versions of GStreamer stuff ***
|
||||
GST_REQ=1.3.1.1
|
||||
GSTPB_REQ=1.3.1.1
|
||||
GSTPG_REQ=1.3.1.1
|
||||
GSTPD_REQ=1.3.1.1
|
||||
GST_REQ=1.3.2
|
||||
GSTPB_REQ=1.3.2
|
||||
GSTPG_REQ=1.3.2
|
||||
GSTPD_REQ=1.3.2
|
||||
|
||||
dnl *** autotools stuff ****
|
||||
|
||||
|
|
|
|||
|
|
@ -28,7 +28,17 @@ RTSP server library based on GStreamer
|
|||
<location rdf:resource="git://anongit.freedesktop.org/gstreamer/gst-rtsp-server"/>
|
||||
<browse rdf:resource="http://cgit.freedesktop.org/gstreamer/gst-rtsp-server"/>
|
||||
</GitRepository>
|
||||
</repository>
|
||||
</repository>
|
||||
|
||||
<release>
|
||||
<Version>
|
||||
<revision>1.3.2</revision>
|
||||
<branch>1.3</branch>
|
||||
<name></name>
|
||||
<created>2014-05-21</created>
|
||||
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.3.2.tar.xz" />
|
||||
</Version>
|
||||
</release>
|
||||
|
||||
<release>
|
||||
<Version>
|
||||
|
|
|
|||
Loading…
Reference in a new issue