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Release 1.7.2
This commit is contained in:
Sebastian Dröge
parent
8f1a9bff7f
commit
60a2fa94b6
5 changed files with 260 additions and 31 deletions
234
ChangeLog
234
ChangeLog
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@ -1,9 +1,237 @@
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=== release 1.7.1 ===
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=== release 1.7.2 ===
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2015-12-24 Sebastian Dröge <slomo@coaxion.net>
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2016-02-19 Sebastian Dröge <slomo@coaxion.net>
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* configure.ac:
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releasing 1.7.1
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releasing 1.7.2
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2016-02-18 15:20:05 +0000 Julien Isorce <j.isorce@samsung.com>
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* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
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uninstalled.pc: add support for non libtool build systems
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Currently the .la path is provided which requires to use libtool as
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mentioned in the GStreamer manual section-helloworld-compilerun.html.
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It is fine as long as the application is built using libtool.
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So currently it is not possible to compile a GStreamer application
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within gst-uninstalled with CMake or other build system different
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than autotools.
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This patch allows to do the following in gst-uninstalled env:
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gcc test.c -o test $(pkg-config --cflags --libs gstreamer-1.0 \
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gstreamer-rtsp-server-1.0)
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Previously it required to prepend libtool --mode=link
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https://bugzilla.gnome.org/show_bug.cgi?id=720778
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2016-02-09 10:34:22 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink: remove check for impossible condition
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Goto error label checks stream to see if it needs to be unreferenced before
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returning, but this goto jumps happens before the stream is ever set, so it
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will always be NULL in this error label.
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CID #1352034
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2016-02-08 23:33:03 +0000 Luis de Bethencourt <luisbg@osg.samsung.com>
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink: clean switch statements
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Coverity demands for fallthrough statements to be clearly commented,
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to distinguish from accidental fall throughs. And it also needs all
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cases to finish with a break, even if the break is never going to be
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executed like in the case of a continue jump.
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CID #1352039
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CID #1352040
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2016-02-05 20:03:01 -0300 Thiago Santos <thiagoss@osg.samsung.com>
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* tests/check/Makefile.am:
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tests: extend the AM_TESTS_ENVIRONMENT from check.mak
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To get the CK_DEFAULT_TIMEOUT defined for all tests
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Also removes a 120 seconds timeout that was set as default
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explicitly in this module
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https://bugzilla.gnome.org/show_bug.cgi?id=761472
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2016-02-05 18:11:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
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* autogen.sh:
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* common:
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Automatic update of common submodule
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From 86e4663 to b64f03f
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2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: fix state_lock not locked again when preroll fails
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https://bugzilla.gnome.org/show_bug.cgi?id=761399
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2016-01-28 22:05:56 +0100 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
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configure: Move plugin specific flags below all the others
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They use some of the other flags, like $GST_ALL_LDFLAGS which is adding
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-no-undefined. And -no-undefined is required on Windows to build DLLs.
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2016-01-28 04:58:00 +1100 Jan Schmidt <jan@centricular.com>
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink: Simplify slightly using new -base API
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Use the new Mikey and SDP API in the base plugins libs
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to simplify some code.
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https://bugzilla.gnome.org/show_bug.cgi?id=758180
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2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
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* .gitignore:
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* configure.ac:
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* gst/Makefile.am:
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* gst/rtsp-sink/Makefile.am:
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* gst/rtsp-sink/gstrtspclientsink.c:
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* gst/rtsp-sink/gstrtspclientsink.h:
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* gst/rtsp-sink/plugin.c:
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* tests/check/Makefile.am:
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* tests/check/gst/rtspclientsink.c:
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rtspsink: Add rtspclientsink element
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Add an rtspclientsink element that accepts streams for which
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there is a registered payloader and sends them to
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an RTSP server using RECORD.
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Sending is synchronised to the pipeline clock. Payload-types
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are automatically selected. The 'new-payloader' signal is fired
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for custom configuration of payloaders when they are created.
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Can now stream a movie like this:
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receiver:
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./test-record "( decodebin name=depay0 ! videoconvert ! autovideosink \
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decodebin name=depay1 ! audioconvert ! autoaudiosink )"
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sender:
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gst-launch-1.0 filesrc location=file-with-aac-and-h264.mp4 ! qtdemux name=d ! \
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queue ! aacparse ! rtspclientsink location=rtsp://127.0.0.1:8554/test name=s \
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https://bugzilla.gnome.org/show_bug.cgi?id=758180
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2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
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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: Add functions for using rtsp-stream from the client
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Add a boolean to indicate that the rtsp-stream is running on the
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'client' side of an RTSP connection, for sending streams via
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RECORD. In that case, the roles of the client/server ports
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in transport setup are swapped.
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https://bugzilla.gnome.org/show_bug.cgi?id=758180
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2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
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* gst/rtsp-server/rtsp-sdp.c:
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* gst/rtsp-server/rtsp-sdp.h:
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rtsp-sdp: Add gst_rtsp_sdp_from_stream()
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A new function that adds info from a GstRTSPStream into an SDP message.
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https://bugzilla.gnome.org/show_bug.cgi?id=758180
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2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Fix mutex beeing unlocked while they should be locked
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https://bugzilla.gnome.org/show_bug.cgi?id=761226
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2016-01-15 07:01:37 +0000 Tim-Philipp Müller <tim@centricular.com>
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* gst/rtsp-server/rtsp-media-factory.c:
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rtsp-media-factory: add missing break in "clock" property setter
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CID 1348453
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2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: fixed assert during update transport
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When RTSP server trying update transport during multicast, it throws an
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assert. The assert is thrown because it is trying to get the parent of
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an non-existing funnel element.
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https://bugzilla.gnome.org/show_bug.cgi?id=760150
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2016-01-03 17:26:31 +0000 Tim-Philipp Müller <tim@centricular.com>
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* gst/rtsp-server/rtsp-permissions.h:
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* gst/rtsp-server/rtsp-thread-pool.h:
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* gst/rtsp-server/rtsp-token.h:
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docs: remove dummy function declarations with G_INLINE_FUNC for gtk-doc
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gtk-doc can handle static inline functions just fine these days,
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there's no need for this stuff any more.
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2015-10-07 18:53:01 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-sdp.c:
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sdp: replace duplicated codes to call new base sdp apis
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https://bugzilla.gnome.org/show_bug.cgi?id=745880
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2015-12-30 16:34:30 +0200 Sebastian Dröge <sebastian@centricular.com>
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* examples/test-netclock.c:
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test-netclock: Use the new API to configure a clock directly
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2015-12-30 16:31:13 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media-factory.c:
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* gst/rtsp-server/rtsp-media-factory.h:
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-media.h:
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rtsp-media: Add API to directly configure a clock on the media pipelines
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2015-12-30 16:43:17 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Fix typo in docs gst_rtsp_media_set_latncy() -> latency()
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2015-12-30 16:30:38 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media-factory.c:
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rtsp-media-factory: Add FIXME for 2.0
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2015-12-30 16:29:45 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Fix indentation
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2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Do not prepare media after media times out
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Deferred calls to start_prepare() can be deferred past the point until
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which wait_preroll() and by proxy gst_rtsp_media_get_status() is
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prepared to wait. Previously there was no lock and no check for this
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situation. This meant that a media could be prepared and unprepared
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simultaneously by two different threads. Now a lock is in place and a
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suitable check is done.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
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2015-12-09 18:24:24 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-media-factory.c:
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* gst/rtsp-server/rtsp-media-factory.h:
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* gst/rtsp-server/rtsp-media.c:
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* gst/rtsp-server/rtsp-media.h:
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rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN
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Without TEARDOWN it might be desireable to keep the media running and continue
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sending data to the client, even if the RTSP connection itself is
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disconnected.
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Only do this for session medias that have only UDP transports. If there's at
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least on TCP transport, it will stop working and cause problems when the
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connection is disconnected.
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https://bugzilla.gnome.org/show_bug.cgi?id=758999
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2015-12-24 15:29:33 +0100 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
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Back to development
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=== release 1.7.1 ===
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2015-12-24 14:54:06 +0100 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 1.7.1
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2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
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2
NEWS
2
NEWS
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@ -1,2 +1,2 @@
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This is GStreamer 1.7.1
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This is GStreamer 1.7.2
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33
RELEASE
33
RELEASE
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@ -1,8 +1,7 @@
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Release notes for GStreamer RTSP Server Library 1.7.1
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Release notes for GStreamer RTSP Server Library 1.7.2
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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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1.7 release series. The 1.7 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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series of the GStreamer multimedia framework. The unstable 1.7 release series
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@ -13,31 +12,26 @@ 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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during the unstable 1.7 release series.
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Bugs fixed in this release
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* 753863 : rtsp-server: examples: Fix memory leaks when context parse fails
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* 756969 : rtsp-server unit tests don't test udp-mcast transport
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* 757949 : gst_rtsp_server_io_func() pops a context that has not been pushed
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* 758179 : GstRTSPStream : Create pipeline based on enabled transport type
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* 758268 : handle_setup_request() expect the media to be suspended
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* 758319 : rtsp-server: Seeking often hangs forever, waiting for prerolling to happen again
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* 758364 : rtsp-session-pool: Avoid dollar sign ($) in session ids
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* 759010 : Valgrind test are faling for rtsp-server for master
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* 758180 : Add rtspclientsink plugin
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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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* 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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You can find source releases of gst-rtsp-server in the download
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directory: http://gstreamer.freedesktop.org/src/gst-rtsp-server/
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directory: https://gstreamer.freedesktop.org/src/gst-rtsp-server/
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The git repository and details how to clone it can be found at
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http://cgit.freedesktop.org/gstreamer/gst-rtsp-server/
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==== Homepage ====
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The project's website is http://gstreamer.freedesktop.org/
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The project's website is https://gstreamer.freedesktop.org/
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==== Support and Bugs ====
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@ -62,17 +56,14 @@ subscribe to the gstreamer-devel list.
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Contributors to this release
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* David Svensson Fors
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* Hyunjun Ko
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* Jan Schmidt
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* Koop Mast
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* Marcus Prebble
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* Nicolas Dufresne
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* Olivier Crête
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* Julien Isorce
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* Luis de Bethencourt
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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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* Vineeth TM
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* Xavier Claessens
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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 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], [1.7.1.1],
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AC_INIT([GStreamer RTSP Server Library], [1.7.2],
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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, 701, 0, 701)
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AS_LIBTOOL(GST, 702, 0, 702)
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dnl *** required versions of GStreamer stuff ***
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GST_REQ=1.7.1.1
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GSTPB_REQ=1.7.1.1
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GSTPG_REQ=1.7.1.1
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GSTPD_REQ=1.7.1.1
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GST_REQ=1.7.2
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GSTPB_REQ=1.7.2
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GSTPG_REQ=1.7.2
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GSTPD_REQ=1.7.2
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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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</repository>
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<release>
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<Version>
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<revision>1.7.2</revision>
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<branch>master</branch>
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<name></name>
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<created>2016-02-19</created>
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<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.7.2.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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<revision>1.7.1</revision>
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