Release 1.7.2

This commit is contained in:
Sebastian Dröge 2016-02-19 12:03:18 +02:00
parent 8f1a9bff7f
commit 60a2fa94b6
5 changed files with 260 additions and 31 deletions

234
ChangeLog
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@ -1,9 +1,237 @@
=== release 1.7.1 ===
=== release 1.7.2 ===
2015-12-24 Sebastian Dröge <slomo@coaxion.net>
2016-02-19 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.7.1
releasing 1.7.2
2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
uninstalled.pc: add support for non libtool build systems
Currently the .la path is provided which requires to use libtool as
mentioned in the GStreamer manual section-helloworld-compilerun.html.
It is fine as long as the application is built using libtool.
So currently it is not possible to compile a GStreamer application
within gst-uninstalled with CMake or other build system different
than autotools.
This patch allows to do the following in gst-uninstalled env:
gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
gstreamer-rtsp-server-1.0)
Previously it required to prepend libtool --mode=link
https://bugzilla.gnome.org/show_bug.cgi?id=720778
2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: remove check for impossible condition
Goto error label checks stream to see if it needs to be unreferenced before
returning, but this goto jumps happens before the stream is ever set, so it
will always be NULL in this error label.
CID #1352034
2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: clean switch statements
Coverity demands for fallthrough statements to be clearly commented,
to distinguish from accidental fall throughs. And it also needs all
cases to finish with a break, even if the break is never going to be
executed like in the case of a continue jump.
CID #1352039
CID #1352040
2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* tests/check/Makefile.am:
tests: extend the AM_TESTS_ENVIRONMENT from check.mak
To get the CK_DEFAULT_TIMEOUT defined for all tests
Also removes a 120 seconds timeout that was set as default
explicitly in this module
https://bugzilla.gnome.org/show_bug.cgi?id=761472
2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
* autogen.sh:
* common:
Automatic update of common submodule
From 86e4663 to b64f03f
2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix state_lock not locked again when preroll fails
https://bugzilla.gnome.org/show_bug.cgi?id=761399
2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
configure: Move plugin specific flags below all the others
They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
-no-undefined. And -no-undefined is required on Windows to build DLLs.
2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Simplify slightly using new -base API
Use the new Mikey and SDP API in the base plugins libs
to simplify some code.
https://bugzilla.gnome.org/show_bug.cgi?id=758180
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* .gitignore:
* configure.ac:
* gst/Makefile.am:
* gst/rtsp-sink/Makefile.am:
* gst/rtsp-sink/gstrtspclientsink.c:
* gst/rtsp-sink/gstrtspclientsink.h:
* gst/rtsp-sink/plugin.c:
* tests/check/Makefile.am:
* tests/check/gst/rtspclientsink.c:
rtspsink: Add rtspclientsink element
Add an rtspclientsink element that accepts streams for which
there is a registered payloader and sends them to
an RTSP server using RECORD.
Sending is synchronised to the pipeline clock. Payload-types
are automatically selected. The 'new-payloader' signal is fired
for custom configuration of payloaders when they are created.
Can now stream a movie like this:
receiver:
./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
decodebin name=depay1 ! audioconvert ! autoaudiosink )"
sender:
gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
https://bugzilla.gnome.org/show_bug.cgi?id=758180
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-stream: Add functions for using rtsp-stream from the client
Add a boolean to indicate that the rtsp-stream is running on the
'client' side of an RTSP connection, for sending streams via
RECORD. In that case, the roles of the client/server ports
in transport setup are swapped.
https://bugzilla.gnome.org/show_bug.cgi?id=758180
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-sdp.c:
* gst/rtsp-server/rtsp-sdp.h:
rtsp-sdp: Add gst_rtsp_sdp_from_stream()
A new function that adds info from a GstRTSPStream into an SDP message.
https://bugzilla.gnome.org/show_bug.cgi?id=758180
2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix mutex beeing unlocked while they should be locked
https://bugzilla.gnome.org/show_bug.cgi?id=761226
2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: add missing break in "clock" property setter
CID 1348453
2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fixed assert during update transport
When RTSP server trying update transport during multicast, it throws an
assert. The assert is thrown because it is trying to get the parent of
an non-existing funnel element.
https://bugzilla.gnome.org/show_bug.cgi?id=760150
2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
* gst/rtsp-server/rtsp-permissions.h:
* gst/rtsp-server/rtsp-thread-pool.h:
* gst/rtsp-server/rtsp-token.h:
docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
gtk-doc can handle static inline functions just fine these days,
there's no need for this stuff any more.
2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-sdp.c:
sdp: replace duplicated codes to call new base sdp apis
https://bugzilla.gnome.org/show_bug.cgi?id=745880
2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/test-netclock.c:
test-netclock: Use the new API to configure a clock directly
2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp-media: Add API to directly configure a clock on the media pipelines
2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: Add FIXME for 2.0
2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Fix indentation
2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Do not prepare media after media times out
Deferred calls to start_prepare() can be deferred past the point until
which wait_preroll() and by proxy gst_rtsp_media_get_status() is
prepared to wait. Previously there was no lock and no check for this
situation. This meant that a media could be prepared and unprepared
simultaneously by two different threads. Now a lock is in place and a
suitable check is done.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media-factory.c:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-media.h:
rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
Without TEARDOWN it might be desireable to keep the media running and continue
sending data to the client, even if the RTSP connection itself is
disconnected.
Only do this for session medias that have only UDP transports. If there's at
least on TCP transport, it will stop working and cause problems when the
connection is disconnected.
https://bugzilla.gnome.org/show_bug.cgi?id=758999
2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.7.1 ===
2015-12-24 14:54:06 +0100 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.7.1
2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>

2
NEWS
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@ -1,2 +1,2 @@
This is GStreamer 1.7.1
This is GStreamer 1.7.2

33
RELEASE
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@ -1,8 +1,7 @@
Release notes for GStreamer RTSP Server Library 1.7.1
Release notes for GStreamer RTSP Server Library 1.7.2
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
1.7 release series. The 1.7 release series is adding new features on top of
the 1.0, 1.2, 1.4 and 1.6 series and is part of the API and ABI-stable 1.x release
series of the GStreamer multimedia framework. The unstable 1.7 release series
@ -15,29 +14,24 @@ during the unstable 1.7 release series.
Bugs fixed in this release
* 753863 : rtsp-server: examples: Fix memory leaks when context parse fails
* 756969 : rtsp-server unit tests don't test udp-mcast transport
* 757949 : gst_rtsp_server_io_func() pops a context that has not been pushed
* 758179 : GstRTSPStream : Create pipeline based on enabled transport type
* 758268 : handle_setup_request() expect the media to be suspended
* 758319 : rtsp-server: Seeking often hangs forever, waiting for prerolling to happen again
* 758364 : rtsp-session-pool: Avoid dollar sign ($) in session ids
* 759010 : Valgrind test are faling for rtsp-server for master
* 758180 : Add rtspclientsink plugin
* 758999 : rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
* 759773 : Prevent simultaneous prepare/unprepare of RTSP media
* 760150 : Updating transport for multicast case gives assertion
==== Download ====
You can find source releases of gst-rtsp-server in the download
directory: http://gstreamer.freedesktop.org/src/gst-rtsp-server/
directory: https://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/
The project's website is https://gstreamer.freedesktop.org/
==== Support and Bugs ====
@ -62,17 +56,14 @@ subscribe to the gstreamer-devel list.
Contributors to this release
* David Svensson Fors
* Hyunjun Ko
* Jan Schmidt
* Koop Mast
* Marcus Prebble
* Nicolas Dufresne
* Olivier Crête
* Julien Isorce
* Luis de Bethencourt
* Sebastian Dröge
* Sebastian Rasmussen
* Srimanta Panda
* Steven Hoving
* Thiago Santos
* Tim-Philipp Müller
* Vineeth TM
* Xavier Claessens
 

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@ -2,7 +2,7 @@ AC_PREREQ(2.69)
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], [1.7.1.1],
AC_INIT([GStreamer RTSP Server Library], [1.7.2],
[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, 701, 0, 701)
AS_LIBTOOL(GST, 702, 0, 702)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.7.1.1
GSTPB_REQ=1.7.1.1
GSTPG_REQ=1.7.1.1
GSTPD_REQ=1.7.1.1
GST_REQ=1.7.2
GSTPB_REQ=1.7.2
GSTPG_REQ=1.7.2
GSTPD_REQ=1.7.2
dnl *** autotools stuff ****

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@ -30,6 +30,16 @@ RTSP server library based on GStreamer
</GitRepository>
</repository>
<release>
<Version>
<revision>1.7.2</revision>
<branch>master</branch>
<name></name>
<created>2016-02-19</created>
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.7.2.tar.xz" />
</Version>
</release>
<release>
<Version>
<revision>1.7.1</revision>