Commit 14d0b77d authored by Tim-Philipp Müller's avatar Tim-Philipp Müller 🐠

Release 1.15.2

parent 7e01dfd1
Pipeline #21487 failed with stages
in 14 minutes and 6 seconds
=== release 1.15.2 ===
2019-02-26 11:58:53 +0000 Tim-Philipp Müller <tim@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
* meson.build:
Release 1.15.2
2019-02-19 09:45:08 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-media.c:
* tests/check/gst/client.c:
rtsp-media: Fix multicast use case with common media
Use case
client 1: SETUP
client 1: PLAY
client 2: SETUP
client 1: TEARDOWN
client 2: PLAY
client 2: TEARDOWN
2019-01-16 12:59:11 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-server/rtsp-stream.h:
rtsp-server: remove recursive behavior
Introduce a threadpool to send rtp and rtcp to avoid recursive behavior.
2019-01-25 14:22:42 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: Only allow to set either a send_func or send_messages_func but not both
And route all messages through the send_func if no send_messages_func
was provided.
We otherwise break backwards compatibility.
2018-09-17 22:18:46 +0300 Sebastian Dröge <sebastian@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-stream.c:
rtsp-client: Add support for sending buffer lists directly
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2018-06-27 12:17:07 +0200 Sebastian Dröge <sebastian@centricular.com>
* docs/libs/gst-rtsp-server-sections.txt:
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-media.c:
* gst/rtsp-server/rtsp-stream-transport.c:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.c:
* gst/rtsp-sink/gstrtspclientsink.c:
rtsp-server: Add support for buffer lists
This adds new functions for passing buffer lists through the different
layers without breaking API/ABI, and enables the appsink to actually
provide buffer lists.
This should already reduce CPU usage and potentially context switches a
bit by passing a whole buffer list from the appsink instead of
individual buffers. As a next step it would be necessary to
a) Add support for a vector of data for the GstRTSPMessage body
b) Add support for sending multiple messages at once to the
GstRTSPWatch and let it be handled internally
c) Adding API to GOutputStream that works like writev()
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2018-12-04 14:12:04 +0100 Benjamin Berg <bberg@redhat.com>
* gst/rtsp-server/rtsp-client.c:
client: Fix crash in close handler
The close handler could trigger a crash because it invalidated the
watch_context while still leaving a source attached to it which would be
cleaned up at a later point.
2019-01-29 14:42:35 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Use cached address when allocating sockets
If an address/port was previously decided upon (ex: multicast in the
SDP), then use that instead of re-creating another one
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/57
2018-12-27 11:28:17 +0100 Lars Wiréen <larswi@axis.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix race codition in finish_unprepare
The previous fix for race condition around finish_unprepare where the
function could be called twice assumed that the status wouldn't change
during execution of the function. This assumption is incorrect as the
state may change, for example if an error message arrives from the
pipeline bus.
Instead a flag keeping track on whether the finish_unprepare function
is currently executing is introduced and checked.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/59
=== release 1.15.1 ===
2019-01-17 02:26:48 +0000 Tim-Philipp Müller <tim@centricular.com>
......
This diff is collapsed.
This is GStreamer gst-rtsp-server 1.15.1.
This is GStreamer gst-rtsp-server 1.15.2.
GStreamer 1.15 is the development branch leading up to the next major
stable version which will be 1.16.
......
......@@ -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], [1.15.1],
AC_INIT([GStreamer RTSP Server Library], [1.15.2],
[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, 1501, 0, 1501)
AS_LIBTOOL(GST, 1502, 0, 1502)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.15.1
GSTPB_REQ=1.15.1
GSTPG_REQ=1.15.1
GSTPD_REQ=1.15.1
GST_REQ=1.15.2
GSTPB_REQ=1.15.2
GSTPG_REQ=1.15.2
GSTPD_REQ=1.15.2
dnl *** autotools stuff ****
......
......@@ -30,6 +30,16 @@ RTSP server library based on GStreamer
</GitRepository>
</repository>
<release>
<Version>
<revision>1.15.2</revision>
<branch>master</branch>
<name></name>
<created>2019-02-26</created>
<file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.15.2.tar.xz" />
</Version>
</release>
<release>
<Version>
<revision>1.15.1</revision>
......
project('gst-rtsp-server', 'c',
version : '1.15.1',
version : '1.15.2',
meson_version : '>= 0.47',
default_options : ['warning_level=1', 'buildtype=debugoptimized'])
......
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