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Release 1.7.90
This commit is contained in:
Sebastian Dröge
parent
0eb3ea03d3
commit
ca1d987a9d
5 changed files with 135 additions and 29 deletions
113
ChangeLog
113
ChangeLog
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=== release 1.7.2 ===
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=== release 1.7.90 ===
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2016-02-19 Sebastian Dröge <slomo@coaxion.net>
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2016-03-01 Sebastian Dröge <slomo@coaxion.net>
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* configure.ac:
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* configure.ac:
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releasing 1.7.2
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releasing 1.7.90
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2016-02-26 12:42:51 +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 b64f03f to 6f2d209
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2016-02-24 00:10:52 +1100 Jan Schmidt <jan@centricular.com>
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* gst/rtsp-sink/gstrtspclientsink.c:
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* tests/check/gst/rtspclientsink.c:
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rtspsink: Fix some leaks in rtspclientsink and the unit test.
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https://bugzilla.gnome.org/show_bug.cgi?id=762525
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2016-02-23 15:01:22 +0100 Patricia Muscalu <patricia@axis.com>
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* tests/check/gst/media.c:
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* tests/check/gst/rtspclientsink.c:
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* tests/check/gst/rtspserver.c:
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* tests/check/gst/stream.c:
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tests: unit test fixes
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Removed port allocation test from the media suite.
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The port allocation failure is now in the stream suite.
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rtspserver:
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Make sure that the media is suspended after the DESCRIBE request
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before reconfiguring the UDP sinks.
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rtspclientsink:
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In the RECORD case we have to set async property to false
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for the appsink element in the test in order to make sure
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that the media pipeline doesn't hang in start_preroll().
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https://bugzilla.gnome.org/show_bug.cgi?id=757488
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2016-02-23 14:59:32 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-stream.c:
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* gst/rtsp-server/rtsp-stream.h:
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rtsp-stream: postpone UDP socket allocation until SETUP
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Postpone the allocation of the UDP sockets until we know
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what transport has been chosen by the client.
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Both unicast and multicast UDP sources are created in one
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function.
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https://bugzilla.gnome.org/show_bug.cgi?id=757488
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2016-01-13 11:29:35 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: postpone the creation of the UDP sources
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Code refactoring: allocate the UDP ports after the sender and
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the reciver parts have been created.
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We postpone the creation of the UDP sources until the UDP
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ports have been allocated.
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https://bugzilla.gnome.org/show_bug.cgi?id=757488
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2016-01-13 10:55:40 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: added function for setting UDP sources to PLAYING state
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Code refactoring: Introduced a function for setting UDP sources
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to PLAYING state.
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https://bugzilla.gnome.org/show_bug.cgi?id=757488
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2015-11-20 15:34:43 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: added function for creating and configuring UDP sources
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Code refactoring: create and configure UDP sources in a separate function.
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https://bugzilla.gnome.org/show_bug.cgi?id=757488
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2015-11-20 14:43:38 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: added function for RTP/RTCP socket configuration
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Code refactoring: configure RTP and RTCP sockets for UDP sinks
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in a separate function.
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https://bugzilla.gnome.org/show_bug.cgi?id=757488
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2015-11-20 08:38:42 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: added function for creating and configuring UDP sinks
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Code refactoring: create and configure UDP sinks in a separate function.
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https://bugzilla.gnome.org/show_bug.cgi?id=757488
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2015-11-19 14:09:25 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: added helper function for creating the sender/receiver parts
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Code refactoring: introduced helper function for creating
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the receiver and the sender parts of the streaming pipeline.
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https://bugzilla.gnome.org/show_bug.cgi?id=757488
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2016-02-19 12:38:42 +0200 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
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Back to development
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=== release 1.7.2 ===
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2016-02-19 12:03:18 +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 1.7.2
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2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
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2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
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2
NEWS
2
NEWS
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@ -1,2 +1,2 @@
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This is GStreamer 1.7.2
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This is GStreamer 1.7.90
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27
RELEASE
27
RELEASE
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@ -1,25 +1,21 @@
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Release notes for GStreamer RTSP Server Library 1.7.2
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Release notes for GStreamer RTSP Server Library 1.7.90
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The GStreamer team is pleased to announce the second release of the unstable
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The GStreamer team is pleased to announce the first release candidate of the stable
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1.7 release series. The 1.7 release series is adding new features on top of
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1.8 release series. The 1.8 release series is adding new features on top of
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the 1.0, 1.2, 1.4 and 1.6 series and is part of the API and ABI-stable 1.x release
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the 1.0, 1.2, 1.4 and 1.6 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 1.7 release series
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series of the GStreamer multimedia framework.
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will lead to the stable 1.8 release series in the next weeks. Any 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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Binaries for Android, iOS, Mac OS X and Windows will be provided separately
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during the unstable 1.7 release series.
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during the stable 1.8 release series.
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Bugs fixed in this release
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Bugs fixed in this release
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* 758180 : Add rtspclientsink plugin
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* 757488 : gst-rtsp-server: multicast functionality is broken if using the same port ranges for multicast and unicast
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* 758999 : rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
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* 762525 : rtspclientsink test: Memory leaks
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* 759773 : Prevent simultaneous prepare/unprepare of RTSP media
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* 760150 : Updating transport for multicast case gives assertion
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==== Download ====
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==== Download ====
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@ -56,14 +52,7 @@ subscribe to the gstreamer-devel list.
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Contributors to this release
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Contributors to this release
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* Hyunjun Ko
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* Jan Schmidt
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* Jan Schmidt
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* Julien Isorce
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* Patricia Muscalu
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* Luis de Bethencourt
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* Sebastian Dröge
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* Sebastian Dröge
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* Sebastian Rasmussen
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* Srimanta Panda
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* Steven Hoving
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* Thiago Santos
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* Tim-Philipp Müller
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12
configure.ac
12
configure.ac
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@ -2,7 +2,7 @@ AC_PREREQ(2.69)
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dnl initialize autoconf
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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 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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dnl releases only do Wall, cvs and prerelease does Werror too
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AC_INIT([GStreamer RTSP Server Library], [1.7.2.1],
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AC_INIT([GStreamer RTSP Server Library], [1.7.90],
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[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
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[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
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[gst-rtsp-server])
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[gst-rtsp-server])
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AG_GST_INIT
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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 1.10.9 (who knows) => 1009
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dnl
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dnl
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dnl sets GST_LT_LDFLAGS
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dnl sets GST_LT_LDFLAGS
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AS_LIBTOOL(GST, 702, 0, 702)
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AS_LIBTOOL(GST, 790, 0, 790)
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dnl *** required versions of GStreamer stuff ***
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dnl *** required versions of GStreamer stuff ***
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GST_REQ=1.7.2.1
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GST_REQ=1.7.90
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GSTPB_REQ=1.7.2.1
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GSTPB_REQ=1.7.90
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GSTPG_REQ=1.7.2.1
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GSTPG_REQ=1.7.90
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GSTPD_REQ=1.7.2.1
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GSTPD_REQ=1.7.90
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dnl *** autotools stuff ****
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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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</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>1.7.90</revision>
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<branch>master</branch>
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<name></name>
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<created>2016-03-01</created>
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<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.7.90.tar.xz" />
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</Version>
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</release>
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<release>
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<release>
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<Version>
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<Version>
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<revision>1.7.2</revision>
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<revision>1.7.2</revision>
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