mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Release 1.7.1
This commit is contained in:
Sebastian Dröge
parent
c934fdaf3b
commit
7374976722
5 changed files with 302 additions and 84 deletions
239
ChangeLog
239
ChangeLog
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@ -1,9 +1,242 @@
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=== release 1.6.0 ===
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=== release 1.7.1 ===
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2015-09-25 Sebastian Dröge <slomo@coaxion.net>
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2015-12-24 Sebastian Dröge <slomo@coaxion.net>
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* configure.ac:
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* configure.ac:
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releasing 1.6.0
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releasing 1.7.1
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2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
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* configure.ac:
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configure: Make -Bsymbolic check work with clang.
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Update the -Bsymbolic check with the version glib has. This version
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works with clang.
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https://bugzilla.gnome.org/show_bug.cgi?id=759713
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2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
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* gst/rtsp-server/rtsp-session-pool.c:
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rtsp-session-pool: Avoid dollar sign ($) in session ids
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Live555 in VLC strips off dollar signs and then gets very confused,
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we don't loose too much entropy by just skipping it.
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2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
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* gst/rtsp-server/rtsp-address-pool.h:
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* gst/rtsp-server/rtsp-auth.h:
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* gst/rtsp-server/rtsp-client.h:
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* gst/rtsp-server/rtsp-media-factory-uri.h:
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* gst/rtsp-server/rtsp-media-factory.h:
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* gst/rtsp-server/rtsp-media.h:
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* gst/rtsp-server/rtsp-mount-points.h:
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* gst/rtsp-server/rtsp-permissions.h:
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* gst/rtsp-server/rtsp-server.h:
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* gst/rtsp-server/rtsp-session-media.h:
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* gst/rtsp-server/rtsp-session-pool.h:
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* gst/rtsp-server/rtsp-session.h:
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* gst/rtsp-server/rtsp-stream-transport.h:
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* gst/rtsp-server/rtsp-stream.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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rtsp-server: Add g_autoptr() support to all types
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https://bugzilla.gnome.org/show_bug.cgi?id=754464
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2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: fixed valgrind error
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Fixed the valgrind error in unit test. The UDP source created during
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gst_rtsp_stream_join_bin() was not released while destroying the rtp
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bin.
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https://bugzilla.gnome.org/show_bug.cgi?id=759010
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2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
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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 b319909 to 86e4663
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2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
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* gst/rtsp-server/rtsp-client.c:
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rtsp-client: suspend media during setup request
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SETUP request from clients needs to suspend the media to clear the
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prerolled buffers. Otherwise it will not affect the prerolled buffer
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and the prerolled buffers will be incorrect (for example block-size
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from setup request will not affect the prerolled buffer unless the
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media is suspended).
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https://bugzilla.gnome.org/show_bug.cgi?id=758268
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2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: create stream pipeline based on transport
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Based on the protocol, create the rtsp stream pipeline. If only TCP or
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only UDP is set as the transport protocol, it will not add the extra tee
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or queue element to the pipeline. Both these elements will be added, if
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it supports both TCP and UDP protocols. This improves the pipeline
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performance when one protocol is present.
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https://bugzilla.gnome.org/show_bug.cgi?id=758179
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2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
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Adding them when not needed will start some logic inside rtpbin that might be
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problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
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would start up a rtpjitterbuffer and behave in weird ways.
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We still set up the UDP sources for RTP receiving for a sender media to be
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able to receive any packets sent by the client for NAT traversal. They will
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all go to a fakesink though.
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Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
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NO_PREROLL, which will cause deadlocks when seeking the media as it will never
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receive ASYNC_DONE after a seek.
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https://bugzilla.gnome.org/show_bug.cgi?id=758319
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2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Disable multicast loopback for the multicast udp sources too
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On POSIX this setting is for sender sockets, on Windows for receiver sockets.
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Previously we were only setting this for sender sockets, which caused looped
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back packets to be received on Windows if a multicast transport was used.
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2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
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* examples/test-record-auth.c:
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* examples/test-record.c:
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examples: Actually use the provided port in the record examples
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2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
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* examples/test-record-auth.c:
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test-record-auth: Add the option to build in TLS support
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2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
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* examples/test-auth.c:
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test-auth: Use an 'anonymous' user for unauthenticated default
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There's a comment on one of the resources that 'user' and 'admin'
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shouldn't even be able to see it, but they can if the default
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token is 'admin2', since that gives them access anyway.
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2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
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* examples/.gitignore:
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* examples/Makefile.am:
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* examples/test-record-auth.c:
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Add test-record-auth example
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2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
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* gst/rtsp-server/rtsp-client.c:
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* tests/check/gst/client.c:
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rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
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2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
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* gst/rtsp-server/rtsp-server.c:
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rtsp-server: Change the logic so we don't pop a NULL context
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When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
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will sometimes fail. This call is made before any context is pushed
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resulting in an attempt to pop a NULL context.
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https://bugzilla.gnome.org/show_bug.cgi?id=757949
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2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
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* tests/check/gst/rtspserver.c:
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rtspserver: Add udp-mcast transport SETUP test
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Refactor utility functions in the test file so they can handle
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more than UDP and TCP as lower transport.
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https://bugzilla.gnome.org/show_bug.cgi?id=756969
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2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
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* gst/rtsp-server/rtsp-stream.c:
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rtsp-stream: Always unref return value of gst_object_get_parent()
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Fixes a leak of a GstBin in the udp-mcast case.
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https://bugzilla.gnome.org/show_bug.cgi?id=756968
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2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
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* common:
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Automatic update of common submodule
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From b99800a to b319909
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2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
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Use new GST_ENABLE_EXTRA_CHECKS #define
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https://bugzilla.gnome.org/show_bug.cgi?id=756870
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2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
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* common:
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Automatic update of common submodule
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From 6babecd to b99800a
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2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
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Update GLib dependency to 2.40.0
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2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
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* examples/test-mp4.c:
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* gst/rtsp-server/rtsp-stream.c:
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stream: listen to sender ssrc signals
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https://bugzilla.gnome.org/show_bug.cgi?id=746747
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2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
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* common:
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common: update for new suppression
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Makes check-valgrind pass with glib 2.46
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2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
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* gst/rtsp-server/rtsp-media.c:
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rtsp-media: Take reference to media that will be prepared
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default_prepare() takes a transfer-none reference GstRTSPMedia object.
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Later on a g_idle_source_new() is created and a pointer to the media
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object is passed as user data. If the media is freed before the idle
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source is dispatched the media object pointer is invalid, but the idle
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source callback expects it to still be valid. To fix this a reference to
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the media object is taken when registering the source callback function
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and a corresponding release of the reference is done when the souce is
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destroyed.
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Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
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2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
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* examples/test-launch.c:
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* examples/test-mp4.c:
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* examples/test-ogg.c:
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* examples/test-record.c:
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* examples/test-uri.c:
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rtsp-server: Fix memory leaks when context parse fails
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When g_option_context_parse fails, context and error variables are not getting free'd
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which results in memory leaks. Free'ing the same.
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And replacing g_error_free with g_clear_error, which checks if the error being passed
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is not NULL and sets the variable to NULL on free'ing.
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https://bugzilla.gnome.org/show_bug.cgi?id=753863
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2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
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* configure.ac:
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Back to development
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=== release 1.6.0 ===
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2015-09-25 23:32:52 +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 1.6.0
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=== release 1.5.91 ===
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=== release 1.5.91 ===
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64
NEWS
64
NEWS
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@ -1,64 +1,2 @@
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This is GStreamer 1.6.0
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This is GStreamer 1.7.1
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The GStreamer team is proud to announce a new major feature release in the
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stable 1.x API series of your favourite cross-platform multimedia framework!
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This release has been in the works for more than a year and is packed with new
|
|
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features, bug fixes and other improvements.
|
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See http://gstreamer.freedesktop.org/releases/1.6/ for the full list of
|
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changes.
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Highlights
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- Stereoscopic 3D and multiview video support
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- Trick mode API for key-frame only fast-forward/fast-reverse playback etc.
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- Improved DTS (decoding timestamp) vs. PTS (presentation timestamp) handling
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to account for negative DTS
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- New GstVideoConverter API for more optimised and more correct conversion of
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raw video frames between all supported formats, with rescaling
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- v4l2src now supports renegotiation
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- v4l2transform can now do scaling
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- V4L2 Element now report Colorimetry properly
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- Easier chunked recording of MP4, Matroska, Ogg, MPEG-TS: new splitmuxsink
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and multifilesink improvements
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- Content Protection signalling API and Common Encryption (CENC) support for
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DASH/MP4
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- Many adaptive streaming (DASH, HLS and MSS) improvements
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- New PTP and NTP network client clocks and better remote clock tracking
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stability
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- High-quality text subtitle overlay at display resolutions with glimagesink
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or gtkglsink
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- RECORD support for the GStreamer RTSP Server
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- Retransmissions (RTX) support in RTSP server and client
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- RTSP seeking support in client and server has been fixed
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- RTCP scheduling improvements and reduced size RTCP support
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- MP4/MOV muxer acquired a new "robust" mode of operation which attempts to
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keep the output file in a valid state at all times
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- Live mixing support in aggregator, audiomixer and compositor was improved a
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lot
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- compositor now supports rescaling and converting inputs streams on the fly
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- New audiointerleave element with proper input synchronisation and live input
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support
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- Blackmagic Design DeckLink capture and playback card support was rewritten
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from scratch; 2k/4k support; mode sensing
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- KLV metadata support in RTP and MPEG-TS
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- H.265 video encoder (x265), decoders (libav, libde265) and RTP payloader and
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depayloaders
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- New DTLS plugin and SRTP/DTLS support
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- OpenGL3 support, multiple contexts and context propagation, 3D video,
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transfer/conversion separation, subtitle blending
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- New OpenGL-based QML video sink, Gtk GL video sink, CoreAnimation
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CAOpenGLLayerSink video sink
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- gst-libav switched to ffmpeg as libav-provider, gains support for
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3D/multiview video, trick modes, and the CAVS codec
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- GstHarness API for unit tests
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- gst-editing-services got a completely new ges-launch-1.0 interface, improved
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mixing support and integration into gst-validate
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- gnonlin has been deprecated in favor of nle (Non Linear Engine) in
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gst-editing-services
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- gst-validate has a new plugin system, an extensive default testsuite,
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support for concurrent test runs and valgrind support
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- cerbero build tool for SDK binary packages gains new 'bundle-source' command
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- Various improvements to the Android, iOS, OS X and Windows platform support
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41
RELEASE
41
RELEASE
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@ -1,24 +1,31 @@
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Release notes for GStreamer RTSP Server Library 1.6.0
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Release notes for GStreamer RTSP Server Library 1.7.1
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The GStreamer team is proud to announce a new major feature release in the
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The GStreamer team is pleased to announce the first release of the unstable
|
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stable 1.x API series of your favourite cross-platform multimedia framework!
|
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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will lead to the stable 1.8 release series in the next weeks. Any newly added
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API can still change until that point.
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This release has been in the works for more than a year and is packed with new
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Binaries for Android, iOS, Mac OS X and Windows will be provided separately
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features, bug fixes and other improvements.
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during the unstable 1.7 release series.
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See
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http://gstreamer.freedesktop.org/releases/1.6/
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|
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for the full list of changes.
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|
|
||||||
There were no bugs fixed in this release
|
Bugs fixed in this release
|
||||||
|
|
||||||
|
* 753863 : rtsp-server: examples: Fix memory leaks when context parse fails
|
||||||
|
* 756969 : rtsp-server unit tests don't test udp-mcast transport
|
||||||
|
* 757949 : gst_rtsp_server_io_func() pops a context that has not been pushed
|
||||||
|
* 758179 : GstRTSPStream : Create pipeline based on enabled transport type
|
||||||
|
* 758268 : handle_setup_request() expect the media to be suspended
|
||||||
|
* 758319 : rtsp-server: Seeking often hangs forever, waiting for prerolling to happen again
|
||||||
|
* 758364 : rtsp-session-pool: Avoid dollar sign ($) in session ids
|
||||||
|
* 759010 : Valgrind test are faling for rtsp-server for master
|
||||||
|
|
||||||
==== Download ====
|
==== Download ====
|
||||||
|
|
||||||
|
|
@ -55,7 +62,17 @@ subscribe to the gstreamer-devel list.
|
||||||
|
|
||||||
Contributors to this release
|
Contributors to this release
|
||||||
|
|
||||||
|
* David Svensson Fors
|
||||||
|
* Hyunjun Ko
|
||||||
* Jan Schmidt
|
* Jan Schmidt
|
||||||
|
* Koop Mast
|
||||||
|
* Marcus Prebble
|
||||||
|
* Nicolas Dufresne
|
||||||
|
* Olivier Crête
|
||||||
* Sebastian Dröge
|
* Sebastian Dröge
|
||||||
|
* Sebastian Rasmussen
|
||||||
|
* Srimanta Panda
|
||||||
* Tim-Philipp Müller
|
* Tim-Philipp Müller
|
||||||
|
* Vineeth TM
|
||||||
|
* Xavier Claessens
|
||||||
|
|
||||||
12
configure.ac
12
configure.ac
|
|
@ -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], [1.7.0.1],
|
AC_INIT([GStreamer RTSP Server Library], [1.7.1],
|
||||||
[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, 700, 0, 700)
|
AS_LIBTOOL(GST, 701, 0, 701)
|
||||||
|
|
||||||
dnl *** required versions of GStreamer stuff ***
|
dnl *** required versions of GStreamer stuff ***
|
||||||
GST_REQ=1.7.0.1
|
GST_REQ=1.7.1
|
||||||
GSTPB_REQ=1.7.0.1
|
GSTPB_REQ=1.7.1
|
||||||
GSTPG_REQ=1.7.0.1
|
GSTPG_REQ=1.7.1
|
||||||
GSTPD_REQ=1.7.0.1
|
GSTPD_REQ=1.7.1
|
||||||
|
|
||||||
dnl *** autotools stuff ****
|
dnl *** autotools stuff ****
|
||||||
|
|
||||||
|
|
|
||||||
|
|
@ -30,6 +30,36 @@ RTSP server library based on GStreamer
|
||||||
</GitRepository>
|
</GitRepository>
|
||||||
</repository>
|
</repository>
|
||||||
|
|
||||||
|
<release>
|
||||||
|
<Version>
|
||||||
|
<revision>1.7.1</revision>
|
||||||
|
<branch>master</branch>
|
||||||
|
<name></name>
|
||||||
|
<created>2015-12-24</created>
|
||||||
|
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.7.1.tar.xz" />
|
||||||
|
</Version>
|
||||||
|
</release>
|
||||||
|
|
||||||
|
<release>
|
||||||
|
<Version>
|
||||||
|
<revision>1.6.2</revision>
|
||||||
|
<branch>1.6</branch>
|
||||||
|
<name></name>
|
||||||
|
<created>2015-12-14</created>
|
||||||
|
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.6.2.tar.xz" />
|
||||||
|
</Version>
|
||||||
|
</release>
|
||||||
|
|
||||||
|
<release>
|
||||||
|
<Version>
|
||||||
|
<revision>1.6.1</revision>
|
||||||
|
<branch>1.6</branch>
|
||||||
|
<name></name>
|
||||||
|
<created>2015-10-30</created>
|
||||||
|
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.6.1.tar.xz" />
|
||||||
|
</Version>
|
||||||
|
</release>
|
||||||
|
|
||||||
<release>
|
<release>
|
||||||
<Version>
|
<Version>
|
||||||
<revision>1.6.0</revision>
|
<revision>1.6.0</revision>
|
||||||
|
|
|
||||||
Loading…
Reference in a new issue