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Release 1.7.1

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Sebastian Dröge 2015-12-24 14:54:06 +01:00
parent c934fdaf3b
commit 7374976722
5 changed files with 302 additions and 84 deletions
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ChangeLog
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@ -1,9 +1,242 @@
=== release 1.6.0 ===
=== release 1.7.1 ===
2015-09-25 Sebastian Dröge <slomo@coaxion.net>
2015-12-24 Sebastian Dröge <slomo@coaxion.net>
* configure.ac:
releasing 1.6.0
releasing 1.7.1
2015-12-21 00:43:49 +0100 Koop Mast <kwm@rainbow-runner.nl>
* configure.ac:
configure: Make -Bsymbolic check work with clang.
Update the -Bsymbolic check with the version glib has. This version
works with clang.
https://bugzilla.gnome.org/show_bug.cgi?id=759713
2015-11-17 22:30:54 -0500 Olivier Crête <olivier.crete@collabora.com>
* gst/rtsp-server/rtsp-session-pool.c:
rtsp-session-pool: Avoid dollar sign ($) in session ids
Live555 in VLC strips off dollar signs and then gets very confused,
we don't loose too much entropy by just skipping it.
2015-11-10 14:17:18 -0500 Xavier Claessens <xavier.claessens@collabora.com>
* gst/rtsp-server/rtsp-address-pool.h:
* gst/rtsp-server/rtsp-auth.h:
* gst/rtsp-server/rtsp-client.h:
* gst/rtsp-server/rtsp-media-factory-uri.h:
* gst/rtsp-server/rtsp-media-factory.h:
* gst/rtsp-server/rtsp-media.h:
* gst/rtsp-server/rtsp-mount-points.h:
* gst/rtsp-server/rtsp-permissions.h:
* gst/rtsp-server/rtsp-server.h:
* gst/rtsp-server/rtsp-session-media.h:
* gst/rtsp-server/rtsp-session-pool.h:
* gst/rtsp-server/rtsp-session.h:
* gst/rtsp-server/rtsp-stream-transport.h:
* gst/rtsp-server/rtsp-stream.h:
* gst/rtsp-server/rtsp-thread-pool.h:
* gst/rtsp-server/rtsp-token.h:
rtsp-server: Add g_autoptr() support to all types
https://bugzilla.gnome.org/show_bug.cgi?id=754464
2015-12-08 08:27:20 +0100 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fixed valgrind error
Fixed the valgrind error in unit test. The UDP source created during
gst_rtsp_stream_join_bin() was not released while destroying the rtp
bin.
https://bugzilla.gnome.org/show_bug.cgi?id=759010
2015-12-07 09:11:35 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
* autogen.sh:
* common:
Automatic update of common submodule
From b319909 to 86e4663
2015-11-18 11:14:39 +0100 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-client.c:
rtsp-client: suspend media during setup request
SETUP request from clients needs to suspend the media to clear the
prerolled buffers. Otherwise it will not affect the prerolled buffer
and the prerolled buffers will be incorrect (for example block-size
from setup request will not affect the prerolled buffer unless the
media is suspended).
https://bugzilla.gnome.org/show_bug.cgi?id=758268
2015-12-04 08:01:37 +0100 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: create stream pipeline based on transport
Based on the protocol, create the rtsp stream pipeline. If only TCP or
only UDP is set as the transport protocol, it will not add the extra tee
or queue element to the pipeline. Both these elements will be added, if
it supports both TCP and UDP protocols. This improves the pipeline
performance when one protocol is present.
https://bugzilla.gnome.org/show_bug.cgi?id=758179
2015-11-19 15:01:16 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Only create RTP sending/receiving rtpbin pads if needed
Adding them when not needed will start some logic inside rtpbin that might be
problematic. Also if e.g. for a sender media we suddenly receive RTP data, we
would start up a rtpjitterbuffer and behave in weird ways.
We still set up the UDP sources for RTP receiving for a sender media to be
able to receive any packets sent by the client for NAT traversal. They will
all go to a fakesink though.
Having an rtpjitterbuffer in the media pipeline will cause the pipeline to be
NO_PREROLL, which will cause deadlocks when seeking the media as it will never
receive ASYNC_DONE after a seek.
https://bugzilla.gnome.org/show_bug.cgi?id=758319
2015-11-17 12:44:38 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Disable multicast loopback for the multicast udp sources too
On POSIX this setting is for sender sockets, on Windows for receiver sockets.
Previously we were only setting this for sender sockets, which caused looped
back packets to be received on Windows if a multicast transport was used.
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* examples/test-record-auth.c:
* examples/test-record.c:
examples: Actually use the provided port in the record examples
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* examples/test-record-auth.c:
test-record-auth: Add the option to build in TLS support
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* examples/test-auth.c:
test-auth: Use an 'anonymous' user for unauthenticated default
There's a comment on one of the resources that 'user' and 'admin'
shouldn't even be able to see it, but they can if the default
token is 'admin2', since that gives them access anyway.
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* examples/.gitignore:
* examples/Makefile.am:
* examples/test-record-auth.c:
Add test-record-auth example
2015-11-17 01:12:28 +1100 Jan Schmidt <jan@centricular.com>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
* gst/rtsp-server/rtsp-server.c:
rtsp-server: Change the logic so we don't pop a NULL context
When doing a port scan (e.g. with nmap) the call to GST_RTSP_CHECK()
will sometimes fail. This call is made before any context is pushed
resulting in an attempt to pop a NULL context.
https://bugzilla.gnome.org/show_bug.cgi?id=757949
2015-10-22 14:32:30 +0200 David Svensson Fors <davidsf@axis.com>
* tests/check/gst/rtspserver.c:
rtspserver: Add udp-mcast transport SETUP test
Refactor utility functions in the test file so they can handle
more than UDP and TCP as lower transport.
https://bugzilla.gnome.org/show_bug.cgi?id=756969
2015-10-22 09:15:21 +0200 David Svensson Fors <davidsf@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: Always unref return value of gst_object_get_parent()
Fixes a leak of a GstBin in the udp-mcast case.
https://bugzilla.gnome.org/show_bug.cgi?id=756968
2015-10-21 14:37:19 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
Automatic update of common submodule
From b99800a to b319909
2015-10-20 17:29:42 +0300 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Use new GST_ENABLE_EXTRA_CHECKS #define
https://bugzilla.gnome.org/show_bug.cgi?id=756870
2015-10-21 14:28:47 +0300 Sebastian Dröge <sebastian@centricular.com>
* common:
Automatic update of common submodule
From 6babecd to b99800a
2015-10-02 22:25:47 +0300 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Update GLib dependency to 2.40.0
2015-10-02 16:11:05 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
* examples/test-mp4.c:
* gst/rtsp-server/rtsp-stream.c:
stream: listen to sender ssrc signals
https://bugzilla.gnome.org/show_bug.cgi?id=746747
2015-09-29 13:00:51 +0100 Tim-Philipp Müller <tim@centricular.com>
* common:
common: update for new suppression
Makes check-valgrind pass with glib 2.46
2015-09-28 17:40:59 +0200 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Take reference to media that will be prepared
default_prepare() takes a transfer-none reference GstRTSPMedia object.
Later on a g_idle_source_new() is created and a pointer to the media
object is passed as user data. If the media is freed before the idle
source is dispatched the media object pointer is invalid, but the idle
source callback expects it to still be valid. To fix this a reference to
the media object is taken when registering the source callback function
and a corresponding release of the reference is done when the souce is
destroyed.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748
2015-08-20 17:01:24 +0900 Vineeth TM <vineeth.tm@samsung.com>
* examples/test-launch.c:
* examples/test-mp4.c:
* examples/test-ogg.c:
* examples/test-record.c:
* examples/test-uri.c:
rtsp-server: Fix memory leaks when context parse fails
When g_option_context_parse fails, context and error variables are not getting free'd
which results in memory leaks. Free'ing the same.
And replacing g_error_free with g_clear_error, which checks if the error being passed
is not NULL and sets the variable to NULL on free'ing.
https://bugzilla.gnome.org/show_bug.cgi?id=753863
2015-09-25 23:51:17 +0200 Sebastian Dröge <sebastian@centricular.com>
* configure.ac:
Back to development
=== release 1.6.0 ===
2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-rtsp-server.doap:
Release 1.6.0
=== release 1.5.91 ===

64
NEWS
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@ -1,64 +1,2 @@
This is GStreamer 1.6.0
The GStreamer team is proud to announce a new major feature release in the
stable 1.x API series of your favourite cross-platform multimedia framework!
This release has been in the works for more than a year and is packed with new
features, bug fixes and other improvements.
See http://gstreamer.freedesktop.org/releases/1.6/ for the full list of
changes.
Highlights
- Stereoscopic 3D and multiview video support
- Trick mode API for key-frame only fast-forward/fast-reverse playback etc.
- Improved DTS (decoding timestamp) vs. PTS (presentation timestamp) handling
to account for negative DTS
- New GstVideoConverter API for more optimised and more correct conversion of
raw video frames between all supported formats, with rescaling
- v4l2src now supports renegotiation
- v4l2transform can now do scaling
- V4L2 Element now report Colorimetry properly
- Easier chunked recording of MP4, Matroska, Ogg, MPEG-TS: new splitmuxsink
and multifilesink improvements
- Content Protection signalling API and Common Encryption (CENC) support for
DASH/MP4
- Many adaptive streaming (DASH, HLS and MSS) improvements
- New PTP and NTP network client clocks and better remote clock tracking
stability
- High-quality text subtitle overlay at display resolutions with glimagesink
or gtkglsink
- RECORD support for the GStreamer RTSP Server
- Retransmissions (RTX) support in RTSP server and client
- RTSP seeking support in client and server has been fixed
- RTCP scheduling improvements and reduced size RTCP support
- MP4/MOV muxer acquired a new "robust" mode of operation which attempts to
keep the output file in a valid state at all times
- Live mixing support in aggregator, audiomixer and compositor was improved a
lot
- compositor now supports rescaling and converting inputs streams on the fly
- New audiointerleave element with proper input synchronisation and live input
support
- Blackmagic Design DeckLink capture and playback card support was rewritten
from scratch; 2k/4k support; mode sensing
- KLV metadata support in RTP and MPEG-TS
- H.265 video encoder (x265), decoders (libav, libde265) and RTP payloader and
depayloaders
- New DTLS plugin and SRTP/DTLS support
- OpenGL3 support, multiple contexts and context propagation, 3D video,
transfer/conversion separation, subtitle blending
- New OpenGL-based QML video sink, Gtk GL video sink, CoreAnimation
CAOpenGLLayerSink video sink
- gst-libav switched to ffmpeg as libav-provider, gains support for
3D/multiview video, trick modes, and the CAVS codec
- GstHarness API for unit tests
- gst-editing-services got a completely new ges-launch-1.0 interface, improved
mixing support and integration into gst-validate
- gnonlin has been deprecated in favor of nle (Non Linear Engine) in
gst-editing-services
- gst-validate has a new plugin system, an extensive default testsuite,
support for concurrent test runs and valgrind support
- cerbero build tool for SDK binary packages gains new 'bundle-source' command
- Various improvements to the Android, iOS, OS X and Windows platform support
This is GStreamer 1.7.1

41
RELEASE
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@ -1,24 +1,31 @@
Release notes for GStreamer RTSP Server Library 1.6.0
Release notes for GStreamer RTSP Server Library 1.7.1
The GStreamer team is proud to announce a new major feature release in the
stable 1.x API series of your favourite cross-platform multimedia framework!
The GStreamer team is pleased to announce the first release of the unstable
1.7 release series. The 1.7 release series is adding new features on top of
the 1.0, 1.2, 1.4 and 1.6 series and is part of the API and ABI-stable 1.x release
series of the GStreamer multimedia framework. The unstable 1.7 release series
will lead to the stable 1.8 release series in the next weeks. Any newly added
API can still change until that point.
This release has been in the works for more than a year and is packed with new
features, bug fixes and other improvements.
See
http://gstreamer.freedesktop.org/releases/1.6/
for the full list of changes.
Binaries for Android, iOS, Mac OS X and Windows will be provided separately
during the unstable 1.7 release series.
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 ====
@ -55,7 +62,17 @@ subscribe to the gstreamer-devel list.
Contributors to this release
* David Svensson Fors
* Hyunjun Ko
* Jan Schmidt
* Koop Mast
* Marcus Prebble
* Nicolas Dufresne
* Olivier Crête
* Sebastian Dröge
* Sebastian Rasmussen
* Srimanta Panda
* Tim-Philipp Müller
* Vineeth TM
* Xavier Claessens
 

12
configure.ac
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@ -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], [1.7.0.1],
AC_INIT([GStreamer RTSP Server Library], [1.7.1],
[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, 700, 0, 700)
AS_LIBTOOL(GST, 701, 0, 701)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.7.0.1
GSTPB_REQ=1.7.0.1
GSTPG_REQ=1.7.0.1
GSTPD_REQ=1.7.0.1
GST_REQ=1.7.1
GSTPB_REQ=1.7.1
GSTPG_REQ=1.7.1
GSTPD_REQ=1.7.1
dnl *** autotools stuff ****

30
gst-rtsp-server.doap
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@ -30,6 +30,36 @@ RTSP server library based on GStreamer
</GitRepository>
</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>
<Version>
<revision>1.6.0</revision>