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Release 1.5.2
This commit is contained in:
Sebastian Dröge
parent
8922afb88d
commit
d8fff9627d
5 changed files with 169 additions and 62 deletions
149
ChangeLog
149
ChangeLog
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@ -1,9 +1,152 @@
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=== release 1.5.1 ===
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=== release 1.5.2 ===
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2015-06-07 Sebastian Dröge <slomo@coaxion.net>
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2015-06-24 Sebastian Dröge <slomo@coaxion.net>
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* configure.ac:
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releasing 1.5.1
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releasing 1.5.2
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2015-06-18 13:12:04 +0200 Ognyan Tonchev <ognyan@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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* gst/rtsp-server/rtsp-client.h:
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* tests/check/gst/client.c:
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rtsp-client: allow application to decide what requirements are supported
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Add "check-requirements" signal and vfunc to allow application
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(and subclasses) to check the requirements.
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Based on patch from Hyunjun Ko <zzoon.ko@samsung.com>
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https://bugzilla.gnome.org/show_bug.cgi?id=749417
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2015-06-16 17:50:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
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* common:
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Automatic update of common submodule
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From 6015d26 to f74b2df
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2015-06-11 17:39:00 +0200 Ognyan Tonchev <ognyan@axis.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Always use real payloader when creating streams
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A bin that contains the real payloader might be used as payloader. In this
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case we have to get the real payloader for the various properties it provides.
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Example use cases for this are bins that payload some media and then have
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additional elements that add metadata or RTP extension headers to the stream.
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https://bugzilla.gnome.org/show_bug.cgi?id=750800
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2015-06-13 17:14:43 +0200 Sebastian Dröge <sebastian@centricular.com>
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* examples/test-netclock-client.c:
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test-netclock: Use gst_pipeline_set_latency() to set a high-enough, equal latency for all receivers
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2015-06-12 23:35:32 +0200 Sebastian Dröge <sebastian@centricular.com>
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* examples/test-netclock-client.c:
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* examples/test-netclock.c:
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test-netclock: Use new ntp-time-source property on rtpbin
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Select the clock time to be used as NTP time source. This allows proper
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synchronization between receivers, independent of sharing base times, and just
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requires them to use the same clock.
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2015-06-11 20:41:31 +0200 Sebastian Dröge <sebastian@centricular.com>
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* examples/test-netclock-client.c:
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* examples/test-netclock.c:
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test-netclock: Setting the same base time on sender and receiver is not necessary
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It's going to be fixed up by rtpbin when using ntp-sync=TRUE
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2015-06-11 17:38:52 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: add description for gst_rtsp_stream_request_aux_sender
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https://bugzilla.gnome.org/show_bug.cgi?id=750764
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2015-06-11 18:10:12 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
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* docs/libs/gst-rtsp-server.types:
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docs: add missing types
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https://bugzilla.gnome.org/show_bug.cgi?id=750764
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2015-06-11 17:37:25 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
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* docs/libs/gst-rtsp-server-sections.txt:
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docs: add missing apis
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https://bugzilla.gnome.org/show_bug.cgi?id=750764
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2015-06-10 17:14:18 +0200 Sebastian Dröge <sebastian@centricular.com>
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* examples/test-netclock-client.c:
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test-netclock-client: Use ntp-sync=TRUE and buffer-mode=SYNC for proper synchronization
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2015-06-05 22:35:39 -0400 Xavier Claessens <xavier.claessens@collabora.com>
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* docs/libs/gst-rtsp-server-sections.txt:
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* gst/rtsp-server/rtsp-auth.c:
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* gst/rtsp-server/rtsp-auth.h:
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GstRTSPAuth: Add client certificate authentication support
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https://bugzilla.gnome.org/show_bug.cgi?id=750471
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2015-06-09 13:53:47 +0200 Sebastian Dröge <sebastian@centricular.com>
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* examples/test-netclock-client.c:
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test-netclock-client: Use new GstClock API to wait for clock synchronization
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2015-06-09 13:51:02 +0200 Sebastian Dröge <sebastian@centricular.com>
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* examples/test-netclock-client.c:
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test-netclock-client: Use a GMainLoop and playbin's source-setup signal
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A mainloop is needed to get glimagesink to display something on OSX, and
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the source-setup signal just makes things a little bit easier.
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2015-06-09 11:30:54 +0200 Edward Hervey <bilboed@bilboed.com>
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* common:
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Automatic update of common submodule
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From d9a3353 to 6015d26
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2015-06-08 23:08:34 +0200 Stefan Sauer <ensonic@users.sf.net>
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* common:
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Automatic update of common submodule
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From d37af32 to d9a3353
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2015-06-07 23:07:31 +0200 Stefan Sauer <ensonic@users.sf.net>
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* common:
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Automatic update of common submodule
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From 21ba2e5 to d37af32
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2015-06-07 17:32:29 +0200 Stefan Sauer <ensonic@users.sf.net>
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* common:
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Automatic update of common submodule
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From c408583 to 21ba2e5
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2015-06-07 17:06:40 +0200 Stefan Sauer <ensonic@users.sf.net>
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* docs/libs/Makefile.am:
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docs: remove variables that we define in the snippet from common
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This is syncing our Makefile.am with upstream gtkdoc.
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2015-06-07 17:16:47 +0200 Stefan Sauer <ensonic@users.sf.net>
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* common:
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Automatic update of common submodule
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From 44a3517 to c408583
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2015-06-07 16:44:55 +0200 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
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Back to development
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=== release 1.5.1 ===
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2015-06-07 11:20:01 +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 1.5.1
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2015-05-25 16:36:18 +0200 Göran Jönsson <goranjn@axis.com>
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2
NEWS
2
NEWS
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@ -1,2 +1,2 @@
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This is GStreamer RTSP Server 1.5.1
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This is GStreamer RTSP Server 1.5.2
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58
RELEASE
58
RELEASE
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@ -1,8 +1,8 @@
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Release notes for GStreamer RTSP Server Library 1.5.1
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Release notes for GStreamer RTSP Server Library 1.5.2
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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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1.5 release series. The 1.5 release series is adding new features on top of
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the 1.0, 1.2 and 1.4 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 1.5 release series
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@ -16,38 +16,11 @@ during the unstable 1.5 release series.
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Features of this release
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Bugs fixed in this release
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* 732238 : Listen on the multicast group for RTP/RTCP packets
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* 734546 : tests: Unref element after usage
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* 736041 : Protect rtsp transport data.
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* 736647 : Tunneled RTSP sessions do not always timeout as expected
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* 737110 : rtsp-client: race condition when closing client connection
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* 737631 : gst-rtsp-server deadlock while sending response over TCP
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* 737675 : media: media_unprepare() is kind of broken
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* 737690 : rtsp-client: deadlock when setting session medias to NULL
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* 737797 : rtsp-stream: lock not released when leaving bin and transports not removed
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* 737829 : rtsp-server: deactivate media when shutting down from paused
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* 738905 : rtsp-client: add stream transport to the context
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* 739112 : rtsp-client: Can not allocate ports for interleaved traffic in setup
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* 740752 : add retransmission support
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* 740845 : crash when reciving a rtcp after teardown but before client finalize.
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* 741678 : configure: add --disable-examples switch
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* 742115 : Examples: Accept a 'port' argument for running multiple instances
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* 742869 : Remove URI-escaping of RTSP session-id
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* 742954 : Crash when two treads are in handle_new_sample at the same time.
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* 743175 : Add support for RECORD
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* 743346 : When system time is increased the ongoing RTSP sessions will time out.
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* 743734 : RTCP packets not sent
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* 744379 : gst-rtsp-server does not preroll when piping data into the media-pipeline
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* 745704 : Losing the first packet
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* 747614 : gst-rtsp-server: uninitialized clock rate causes critical warning
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* 747839 : gst-rtsp-server: doesn't perform retransmission to both streams in test-video-rtx
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* 748058 : autogen.sh fails due to autopoint erroring out due to missing gettext version in configure.ac
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* 749845 : Client have problem to find the teardown response.
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* 749417 : rtsp-client: add API to allow application to decide what requirements are supported
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* 750764 : gst-rtsp-server: add missing apis to doc
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* 750800 : rtsp-media: always use real payloader when creating streams
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==== Download ====
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@ -86,30 +59,11 @@ Applications
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Contributors to this release
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* Aleix Conchillo Flaqué
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* Alistair Buxton
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* Andreas Frisch
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* Anila Balavan
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* Arun Raghavan
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* Branko Subasic
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* Edward Hervey
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* Gregor Boirie
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* Göran Jönsson
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* Hyunjun Ko
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* Jan Schmidt
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* Kent-Inge Ingesson
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* Linus Svensson
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* Luis de Bethencourt
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* Matthew Waters
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* Nicolas Dufresne
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* Nirbheek Chauhan
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* Ognyan Tonchev
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* Olivier Crête
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* Sebastian Dröge
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* Sebastian Rasmussen
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* Srimanta Panda
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* Stefan Sauer
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* Tim-Philipp Müller
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* Vincent Penquerc'h
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* Wim Taymans
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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], [1.5.1.1],
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AC_INIT([GStreamer RTSP Server Library], [1.5.2],
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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, 501, 0, 501)
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AS_LIBTOOL(GST, 502, 0, 502)
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dnl *** required versions of GStreamer stuff ***
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GST_REQ=1.5.1.1
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GSTPB_REQ=1.5.1.1
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GSTPG_REQ=1.5.1.1
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GSTPD_REQ=1.5.1.1
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GST_REQ=1.5.2
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GSTPB_REQ=1.5.2
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GSTPG_REQ=1.5.2
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GSTPD_REQ=1.5.2
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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>1.5.2</revision>
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<branch>1.5</branch>
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<name></name>
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<created>2015-06-24</created>
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<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.5.2.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>1.5.1</revision>
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