mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-29 21:21:12 +00:00
Code Issues 1 Projects Releases Wiki Activity

Release 1.11.1

Browse source
This commit is contained in:
Sebastian Dröge 2017-01-12 16:14:46 +02:00
parent fb7833245d
commit c860590a8c
6 changed files with 293 additions and 1136 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

244
ChangeLog
View file

@ -1,9 +1,247 @@
=== release 1.10.0 === === release 1.11.1 ===
2016-11-01 Sebastian Dröge <slomo@coaxion.net> 2017-01-12 Sebastian Dröge <slomo@coaxion.net>
* configure.ac: * configure.ac:
releasing 1.10.0 releasing 1.11.1
2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: corrected if-statement in _get_server_port()
This bug was accidentally introduced while fixing a segfault
in _get_server_port() function.
https://bugzilla.gnome.org/show_bug.cgi?id=776345
2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/stream.c:
rtsp-stream: fixed segmenation fault in _get_server_port()
Calling function gst_rtsp_stream_get_server_port() results in
segmenation fault in the RTP/RTSP/TCP case.
Port that the server will use to receive RTCP makes only
sense in the UDP case, however the function should handle
the TCP case in a nicer way.
https://bugzilla.gnome.org/show_bug.cgi?id=776345
2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
* gst/rtsp-server/rtsp-media-factory.c:
dosc: Fix a little typo
https://bugzilla.gnome.org/show_bug.cgi?id=777037
2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
* pkgconfig/meson.build:
meson: generate pkg-config -uninstalled pc files
Generating those files is useful for users building the GStreamer stack
using meson and having to link it to another project which is still
using the autotools.
https://bugzilla.gnome.org/show_bug.cgi?id=776810
2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
pkgconfig: fix -uninstalled pc file
pcfiledir was never defined so the paths were wrong.
https://bugzilla.gnome.org/show_bug.cgi?id=776867
2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
* gst/rtsp-server/rtsp-stream.c:
* tests/check/gst/rtspserver.c:
rtsp-stream: Fixed TCP transport case
Make sure that the appsink element is actually added to
the bin before trying to link it with the elements in it.
https://bugzilla.gnome.org/show_bug.cgi?id=776343
2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
* Makefile.am:
* configure.ac:
* gst-rtsp.spec.in:
Remove generated .spec file
Likely extremely bitrotten, and we should not ship this anyway.
2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
* common:
Automatic update of common submodule
From f980fd9 to 39ac2f5
2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media.c:
media: Fix pt map caps
Since decryption is handled within rtpbin, all outcoming stream
caps will be application/x-rtp (i.e. regular rtp)
Fixes RECORD with SRTP streams
2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
media-factory: Create media objects with the proper transport mode
The function called immediately afterwards (collect_streams()) will
need it to work properly
2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-auth.c:
rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-media-factory.c:
rtsp-media-factory: Don't create a pipeline for the media pipeline string
We're going to put a pipeline into a pipeline otherwise, which is not
exactly ideal.
2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
* gst/rtsp-server/rtsp-media.c:
media: Fix race condition around finish_unprepare() if called multiple time
https://bugzilla.gnome.org/show_bug.cgi?id=755329
2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Don't leave stale pointer after unref
Fix a warning on shutdown - don't keep a pointer to an
alread-unreffed object.
2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
* .gitmodules:
common: use https protocol for common submodule
https://bugzilla.gnome.org/show_bug.cgi?id=775110
2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
stream: block the output of rtpbin instead of the source pipeline
85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
detection of the srtp rollover counter to add to the SDP.
Unfortunately, it was incomplete for live pipelines where the logic
blocks the source bin before creating the SDP and thus would never have
the necessary informaiton to create a correct SDP with srtp encryption.
Move the pad blocks to rtpbin's output pads instead so that the
necessary information can be created before we need the information for
the SDP.
https://bugzilla.gnome.org/show_bug.cgi?id=770239
2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: add IDLE timeout, before session exists
The RTSP server will not timeout an idle RTSP connection
(note this is different from doing timeout on a RTSP
session).
At least for Apache this is a problem when running RTSP over
HTTPS since it uses one of the threads (there is a rather
limited number) that are available for handling requests.
https://bugzilla.gnome.org/show_bug.cgi?id=771830
2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
* .gitignore:
.gitignore more
2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Set close-socket FALSE on UDP src:es
With this RTSP server can use the sockets independent on the udpsrc
state.
When the udp src is finalized it will unref socket and when g_socket
is finalized the socket will be closed.
https://bugzilla.gnome.org/show_bug.cgi?id=765673
2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-sink/gstrtspclientsink.c:
rtspclientsink: Move to new helper function to parse authentication responses
https://bugzilla.gnome.org/show_bug.cgi?id=774416
2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
* examples/Makefile.am:
* examples/test-auth-digest.c:
* gst/rtsp-server/rtsp-auth.c:
* gst/rtsp-server/rtsp-auth.h:
* win32/common/libgstrtspserver.def:
rtsp-auth: Add support for Digest authentication
https://bugzilla.gnome.org/show_bug.cgi?id=774416
2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
* Makefile.am:
* gst/rtsp-server/meson.build:
* meson.build:
* tests/check/meson.build:
* win32/MANIFEST:
* win32/common/libgstrtspserver.def:
Enable building with MSVC
https://bugzilla.gnome.org/show_bug.cgi?id=774640
2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
* meson.build:
meson: gstreamer gst_check_dep does not exist on windows
2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
* gst/rtsp-server/rtsp-client.c:
client: update do_send_message to match type GstRTSPClientSendFunc
This type mismatch fails building with MSVC
https://bugzilla.gnome.org/show_bug.cgi?id=774640
2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-sdp.c:
rtsp-sdp: Fix indentation
2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Only signal "new-state" if the state has actually changed
https://bugzilla.gnome.org/show_bug.cgi?id=774173
2016-08-24 11:39:13 +0200 Branko Subasic <branko@axis.com>
* gst/rtsp-server/rtsp-client.c:
* gst/rtsp-server/rtsp-client.h:
client: emit signal in the beginning of each rtsp request
These signals let the application validate the requests, configure the
media/stream in a certain way and also generate error status code in
case of error or bad request.
https://bugzilla.gnome.org/show_bug.cgi?id=758062
2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
* meson.build:
meson: update version
=== release 1.11.0 ===
2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.10.0 ===
2016-11-01 18:06:46 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.10.0
2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com> 2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>

1115
NEWS
View file

File diff suppressed because it is too large Load diff

46
RELEASE
View file

@ -1,23 +1,32 @@
Release notes for GStreamer RTSP Server Library 1.10.0 Release notes for GStreamer RTSP Server Library 1.11.1
The GStreamer team is pleased to announce the first release of the new stable The GStreamer team is pleased to announce the first release of the unstable
1.10 release series. The 1.10 release series is adding new features on top of 1.11 release series. The 1.11 release series is adding new features on top of
the 1.0, 1.2, 1.4, 1.6 and 1.8 series and is part of the API and ABI-stable 1.x the 1.0, 1.2, 1.4, 1.6, 1.8 and 1.10 series and is part of the API and ABI-stable 1.x release
release series of the GStreamer multimedia framework. series of the GStreamer multimedia framework. The unstable 1.11 release series
will lead to the stable 1.12 release series in the next weeks. Any newly added
API can still change until that point.
Binaries for Android, iOS, Mac OS X and Windows will be provided shortly after Full release notes will be provided at some point during the 1.11 release
the source release by the GStreamer project during the stable 1.10 release cycle, highlighting all the new features, bugfixes, performance optimizations
series. and other important changes.
Binaries for Android, iOS, Mac OS X and Windows will be provided in the next days.
Bugs fixed in this release Bugs fixed in this release
* 771983 : Deadlock when closing session and backlog is full. * 758062 : rtsp-client: emit new rtsp request signals in the beginning of each request
* 772478 : Missing video stream from SDP * 771830 : There is no time out in idle connection RTSP server
* 773640 : rtspclient unit test failures * 774173 : media: emit signal SIGNAL_NEW_STATE only when state change happens
* 774640 : gst-rtsp-server: Enable building with MSVC
* 776867 : pkgconfig: fix -uninstalled pc file
* 777037 : rtsp-factory: just fixing a little typo in comments
* 774416 : RTSP digest Authentification for gst-rtsp-server
==== Download ==== ==== Download ====
@ -54,8 +63,19 @@ subscribe to the gstreamer-devel list.
Contributors to this release Contributors to this release
* Aleksandr Slobodeniuk
* Branko Subasic
* Dag Gullberg
* Edward Hervey
* Guillaume Desmottes
* Göran Jönsson * Göran Jönsson
* Nikita Bobkov * Jan Schmidt
* Kseniia Vasilchuk
* Matthew Waters
* Neha Arora
* Patricia Muscalu
* Scott D Phillips
* Sebastian Dröge
* Thibault Saunier
* Tim-Philipp Müller * Tim-Philipp Müller
* Xavier Claessens
   

12
configure.ac
View file

@ -2,7 +2,7 @@ AC_PREREQ(2.69)
dnl initialize autoconf dnl initialize autoconf
dnl when going to/from release please set the nano (fourth number) right ! dnl when going to/from release please set the nano (fourth number) right !
dnl releases only do Wall, cvs and prerelease does Werror too dnl releases only do Wall, cvs and prerelease does Werror too
AC_INIT([GStreamer RTSP Server Library], [1.11.0.1], AC_INIT([GStreamer RTSP Server Library], [1.11.1],
[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer], [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
[gst-rtsp-server]) [gst-rtsp-server])
AG_GST_INIT AG_GST_INIT
@ -53,13 +53,13 @@ dnl 1.2.5 => 205
dnl 1.10.9 (who knows) => 1009 dnl 1.10.9 (who knows) => 1009
dnl dnl
dnl sets GST_LT_LDFLAGS dnl sets GST_LT_LDFLAGS
AS_LIBTOOL(GST, 1100, 0, 1100) AS_LIBTOOL(GST, 1101, 0, 1101)
dnl *** required versions of GStreamer stuff *** dnl *** required versions of GStreamer stuff ***
GST_REQ=1.11.0.1 GST_REQ=1.11.1
GSTPB_REQ=1.11.0.1 GSTPB_REQ=1.11.1
GSTPG_REQ=1.11.0.1 GSTPG_REQ=1.11.1
GSTPD_REQ=1.11.0.1 GSTPD_REQ=1.11.1
dnl *** autotools stuff **** dnl *** autotools stuff ****

10
gst-rtsp-server.doap
View file

@ -30,6 +30,16 @@ RTSP server library based on GStreamer
</GitRepository> </GitRepository>
</repository> </repository>
<release>
<Version>
<revision>1.11.1</revision>
<branch>master</branch>
<name></name>
<created>2017-01-12</created>
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.11.1.tar.xz" />
</Version>
</release>
<release> <release>
<Version> <Version>
<revision>1.10.0</revision> <revision>1.10.0</revision>

2
win32/common/libgstrtspserver.def
View file

@ -197,6 +197,7 @@ EXPORTS
gst_rtsp_session_get_sessionid gst_rtsp_session_get_sessionid
gst_rtsp_session_get_timeout gst_rtsp_session_get_timeout
gst_rtsp_session_get_type gst_rtsp_session_get_type
gst_rtsp_session_is_expired
gst_rtsp_session_is_expired_usec gst_rtsp_session_is_expired_usec
gst_rtsp_session_manage_media gst_rtsp_session_manage_media
gst_rtsp_session_media_alloc_channels gst_rtsp_session_media_alloc_channels
@ -212,6 +213,7 @@ EXPORTS
gst_rtsp_session_media_set_state gst_rtsp_session_media_set_state
gst_rtsp_session_media_set_transport gst_rtsp_session_media_set_transport
gst_rtsp_session_new gst_rtsp_session_new
gst_rtsp_session_next_timeout
gst_rtsp_session_next_timeout_usec gst_rtsp_session_next_timeout_usec
gst_rtsp_session_pool_cleanup gst_rtsp_session_pool_cleanup
gst_rtsp_session_pool_create gst_rtsp_session_pool_create