mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Release 1.11.1
This commit is contained in:
Sebastian Dröge
parent
fb7833245d
commit
c860590a8c
6 changed files with 293 additions and 1136 deletions
244
ChangeLog
244
ChangeLog
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@ -1,9 +1,247 @@
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=== release 1.10.0 ===
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=== release 1.11.1 ===
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2016-11-01 Sebastian Dröge <slomo@coaxion.net>
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2017-01-12 Sebastian Dröge <slomo@coaxion.net>
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* configure.ac:
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releasing 1.10.0
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releasing 1.11.1
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2017-01-10 08:34:50 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: corrected if-statement in _get_server_port()
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This bug was accidentally introduced while fixing a segfault
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in _get_server_port() function.
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https://bugzilla.gnome.org/show_bug.cgi?id=776345
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2017-01-09 14:12:05 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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* tests/check/gst/stream.c:
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rtsp-stream: fixed segmenation fault in _get_server_port()
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Calling function gst_rtsp_stream_get_server_port() results in
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segmenation fault in the RTP/RTSP/TCP case.
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Port that the server will use to receive RTCP makes only
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sense in the UDP case, however the function should handle
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the TCP case in a nicer way.
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https://bugzilla.gnome.org/show_bug.cgi?id=776345
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2017-01-09 12:22:40 +0300 Aleksandr Slobodeniuk <alenuke@yandex.ru>
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* gst/rtsp-server/rtsp-media-factory.c:
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dosc: Fix a little typo
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https://bugzilla.gnome.org/show_bug.cgi?id=777037
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2017-01-04 16:20:54 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
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* pkgconfig/Makefile.am:
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* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
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* pkgconfig/meson.build:
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meson: generate pkg-config -uninstalled pc files
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Generating those files is useful for users building the GStreamer stack
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using meson and having to link it to another project which is still
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using the autotools.
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https://bugzilla.gnome.org/show_bug.cgi?id=776810
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2017-01-04 16:11:08 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
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* pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in:
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pkgconfig: fix -uninstalled pc file
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pcfiledir was never defined so the paths were wrong.
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https://bugzilla.gnome.org/show_bug.cgi?id=776867
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2016-12-21 13:41:50 +0100 Patricia Muscalu <patricia@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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* tests/check/gst/rtspserver.c:
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rtsp-stream: Fixed TCP transport case
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Make sure that the appsink element is actually added to
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the bin before trying to link it with the elements in it.
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https://bugzilla.gnome.org/show_bug.cgi?id=776343
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2016-12-16 17:26:04 +0000 Tim-Philipp Müller <tim@centricular.com>
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* .gitignore:
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* Makefile.am:
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* configure.ac:
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* gst-rtsp.spec.in:
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Remove generated .spec file
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Likely extremely bitrotten, and we should not ship this anyway.
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2016-12-03 08:21:02 +0100 Edward Hervey <bilboed@bilboed.com>
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* common:
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Automatic update of common submodule
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From f980fd9 to 39ac2f5
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2016-12-02 15:40:09 +0100 Edward Hervey <edward@centricular.com>
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* gst/rtsp-server/rtsp-media.c:
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media: Fix pt map caps
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Since decryption is handled within rtpbin, all outcoming stream
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caps will be application/x-rtp (i.e. regular rtp)
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Fixes RECORD with SRTP streams
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2016-12-02 15:38:04 +0100 Edward Hervey <edward@centricular.com>
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* gst/rtsp-server/rtsp-media-factory.c:
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media-factory: Create media objects with the proper transport mode
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The function called immediately afterwards (collect_streams()) will
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need it to work properly
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2016-12-02 14:36:50 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-auth.c:
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rtsp-auth: Don't remove digest-auth nonces that already/still have a client connected
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2016-12-01 18:04:34 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-media-factory.c:
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rtsp-media-factory: Don't create a pipeline for the media pipeline string
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We're going to put a pipeline into a pipeline otherwise, which is not
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exactly ideal.
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2016-10-25 15:41:28 +0300 Kseniia Vasilchuk <vasilchukkseniia@gmail.com>
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* gst/rtsp-server/rtsp-media.c:
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media: Fix race condition around finish_unprepare() if called multiple time
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https://bugzilla.gnome.org/show_bug.cgi?id=755329
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2016-11-30 14:06:36 +1100 Jan Schmidt <jan@centricular.com>
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink: Don't leave stale pointer after unref
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Fix a warning on shutdown - don't keep a pointer to an
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alread-unreffed object.
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2016-11-26 11:24:50 +0000 Tim-Philipp Müller <tim@centricular.com>
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* .gitmodules:
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common: use https protocol for common submodule
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https://bugzilla.gnome.org/show_bug.cgi?id=775110
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2016-11-21 23:29:56 +1100 Matthew Waters <matthew@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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stream: block the output of rtpbin instead of the source pipeline
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85c52e194bcb81928b96614be0ae47d59eccb1ce introduced a more correct
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detection of the srtp rollover counter to add to the SDP.
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Unfortunately, it was incomplete for live pipelines where the logic
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blocks the source bin before creating the SDP and thus would never have
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the necessary informaiton to create a correct SDP with srtp encryption.
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Move the pad blocks to rtpbin's output pads instead so that the
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necessary information can be created before we need the information for
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the SDP.
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https://bugzilla.gnome.org/show_bug.cgi?id=770239
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2016-11-21 16:02:39 +0100 Dag Gullberg <dagg@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: add IDLE timeout, before session exists
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The RTSP server will not timeout an idle RTSP connection
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(note this is different from doing timeout on a RTSP
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session).
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At least for Apache this is a problem when running RTSP over
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HTTPS since it uses one of the threads (there is a rather
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limited number) that are available for handling requests.
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https://bugzilla.gnome.org/show_bug.cgi?id=771830
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2016-11-23 09:45:08 +0000 Tim-Philipp Müller <tim@centricular.com>
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* .gitignore:
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.gitignore more
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2016-11-21 13:05:50 +0100 Göran Jönsson <goranjn@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Set close-socket FALSE on UDP src:es
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With this RTSP server can use the sockets independent on the udpsrc
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state.
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When the udp src is finalized it will unref socket and when g_socket
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is finalized the socket will be closed.
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https://bugzilla.gnome.org/show_bug.cgi?id=765673
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2016-11-18 17:47:13 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-sink/gstrtspclientsink.c:
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rtspclientsink: Move to new helper function to parse authentication responses
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https://bugzilla.gnome.org/show_bug.cgi?id=774416
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2016-11-16 08:42:24 +0200 Sebastian Dröge <sebastian@centricular.com>
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* examples/Makefile.am:
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* examples/test-auth-digest.c:
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* gst/rtsp-server/rtsp-auth.c:
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* gst/rtsp-server/rtsp-auth.h:
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* win32/common/libgstrtspserver.def:
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rtsp-auth: Add support for Digest authentication
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https://bugzilla.gnome.org/show_bug.cgi?id=774416
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2016-11-17 09:41:53 -0800 Scott D Phillips <scott.d.phillips@intel.com>
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* Makefile.am:
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* gst/rtsp-server/meson.build:
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* meson.build:
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* tests/check/meson.build:
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* win32/MANIFEST:
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* win32/common/libgstrtspserver.def:
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Enable building with MSVC
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https://bugzilla.gnome.org/show_bug.cgi?id=774640
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2016-11-18 20:23:14 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
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* meson.build:
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meson: gstreamer gst_check_dep does not exist on windows
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2016-11-17 09:43:37 -0800 Scott D Phillips <scott.d.phillips@intel.com>
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* gst/rtsp-server/rtsp-client.c:
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client: update do_send_message to match type GstRTSPClientSendFunc
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This type mismatch fails building with MSVC
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https://bugzilla.gnome.org/show_bug.cgi?id=774640
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2016-11-11 14:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-sdp.c:
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rtsp-sdp: Fix indentation
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2016-11-10 05:16:00 +0000 Neha Arora <arora.neha@samsung.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Only signal "new-state" if the state has actually changed
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https://bugzilla.gnome.org/show_bug.cgi?id=774173
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2016-08-24 11:39:13 +0200 Branko Subasic <branko@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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client: emit signal in the beginning of each rtsp request
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These signals let the application validate the requests, configure the
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media/stream in a certain way and also generate error status code in
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case of error or bad request.
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https://bugzilla.gnome.org/show_bug.cgi?id=758062
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2016-11-01 18:10:35 +0000 Tim-Philipp Müller <tim@centricular.com>
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* meson.build:
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meson: update version
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=== release 1.11.0 ===
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2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
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Back to development
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=== release 1.10.0 ===
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2016-11-01 18:06:46 +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 1.10.0
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2016-10-28 18:38:01 +0100 Tim-Philipp Müller <tim@centricular.com>
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46
RELEASE
46
RELEASE
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@ -1,23 +1,32 @@
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Release notes for GStreamer RTSP Server Library 1.10.0
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Release notes for GStreamer RTSP Server Library 1.11.1
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The GStreamer team is pleased to announce the first release of the new stable
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1.10 release series. The 1.10 release series is adding new features on top of
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the 1.0, 1.2, 1.4, 1.6 and 1.8 series and is part of the API and ABI-stable 1.x
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release series of the GStreamer multimedia framework.
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The GStreamer team is pleased to announce the first release of the unstable
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1.11 release series. The 1.11 release series is adding new features on top of
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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
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series of the GStreamer multimedia framework. The unstable 1.11 release series
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will lead to the stable 1.12 release series in the next weeks. Any newly added
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API can still change until that point.
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Binaries for Android, iOS, Mac OS X and Windows will be provided shortly after
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the source release by the GStreamer project during the stable 1.10 release
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series.
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Full release notes will be provided at some point during the 1.11 release
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cycle, highlighting all the new features, bugfixes, performance optimizations
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and other important changes.
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Binaries for Android, iOS, Mac OS X and Windows will be provided in the next days.
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Bugs fixed in this release
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* 771983 : Deadlock when closing session and backlog is full.
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* 772478 : Missing video stream from SDP
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* 773640 : rtspclient unit test failures
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* 758062 : rtsp-client: emit new rtsp request signals in the beginning of each request
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* 771830 : There is no time out in idle connection RTSP server
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* 774173 : media: emit signal SIGNAL_NEW_STATE only when state change happens
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* 774640 : gst-rtsp-server: Enable building with MSVC
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* 776867 : pkgconfig: fix -uninstalled pc file
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* 777037 : rtsp-factory: just fixing a little typo in comments
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* 774416 : RTSP digest Authentification for gst-rtsp-server
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==== Download ====
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@ -54,8 +63,19 @@ subscribe to the gstreamer-devel list.
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Contributors to this release
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* Aleksandr Slobodeniuk
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* Branko Subasic
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* Dag Gullberg
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* Edward Hervey
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* Guillaume Desmottes
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* Göran Jönsson
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* Nikita Bobkov
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* Jan Schmidt
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* Kseniia Vasilchuk
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* Matthew Waters
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* Neha Arora
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* Patricia Muscalu
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* Scott D Phillips
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* Sebastian Dröge
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* Thibault Saunier
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* Tim-Philipp Müller
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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], [1.11.0.1],
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AC_INIT([GStreamer RTSP Server Library], [1.11.1],
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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, 1100, 0, 1100)
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AS_LIBTOOL(GST, 1101, 0, 1101)
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dnl *** required versions of GStreamer stuff ***
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GST_REQ=1.11.0.1
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GSTPB_REQ=1.11.0.1
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GSTPG_REQ=1.11.0.1
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GSTPD_REQ=1.11.0.1
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GST_REQ=1.11.1
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GSTPB_REQ=1.11.1
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GSTPG_REQ=1.11.1
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GSTPD_REQ=1.11.1
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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>1.11.1</revision>
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<branch>master</branch>
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<name></name>
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<created>2017-01-12</created>
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<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.11.1.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>1.10.0</revision>
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@ -197,6 +197,7 @@ EXPORTS
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gst_rtsp_session_get_sessionid
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gst_rtsp_session_get_timeout
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gst_rtsp_session_get_type
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gst_rtsp_session_is_expired
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gst_rtsp_session_is_expired_usec
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gst_rtsp_session_manage_media
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gst_rtsp_session_media_alloc_channels
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@ -212,6 +213,7 @@ EXPORTS
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gst_rtsp_session_media_set_state
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gst_rtsp_session_media_set_transport
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gst_rtsp_session_new
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gst_rtsp_session_next_timeout
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gst_rtsp_session_next_timeout_usec
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gst_rtsp_session_pool_cleanup
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gst_rtsp_session_pool_create
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