Commit 5e42fd5a authored by Tim-Philipp Müller's avatar Tim-Philipp Müller 🐠

Release 1.6.4

parent 3165ff37
=== release 1.6.4 ===
2016-04-14 Tim-Philipp Müller <tim@centricular.com>
* configure.ac:
releasing 1.6.4
2016-01-05 13:10:36 +0100 Srimanta Panda <srimanta@axis.com>
* gst/rtsp-server/rtsp-stream.c:
rtsp-stream: fixed assert during update transport
When RTSP server trying update transport during multicast, it throws an
assert. The assert is thrown because it is trying to get the parent of
an non-existing funnel element.
https://bugzilla.gnome.org/show_bug.cgi?id=760150
2015-12-22 12:08:02 +0100 Sebastian Rasmussen <sebras@hotmail.com>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Do not prepare media after media times out
Deferred calls to start_prepare() can be deferred past the point until
which wait_preroll() and by proxy gst_rtsp_media_get_status() is
prepared to wait. Previously there was no lock and no check for this
situation. This meant that a media could be prepared and unprepared
simultaneously by two different threads. Now a lock is in place and a
suitable check is done.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759773
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.
2015-12-08 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
2015-11-18 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
2015-11-19 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>
* gst/rtsp-server/rtsp-client.c:
* tests/check/gst/client.c:
rtsp-client: Report RECORD and ANNOUNCE as supported in the OPTIONS
2016-02-02 09:01:51 +0100 Steven Hoving <sh@bigbrother.nl>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: fix state_lock not locked again when preroll fails
https://bugzilla.gnome.org/show_bug.cgi?id=761399
2016-01-28 09:22:18 +0100 Steven Hoving <sh@bigbrother.nl>
* gst/rtsp-server/rtsp-media.c:
rtsp-media: Fix mutex beeing unlocked while they should be locked
https://bugzilla.gnome.org/show_bug.cgi?id=761226
=== release 1.6.2 ===
2015-12-14 Sebastian Dröge <slomo@coaxion.net>
2015-12-14 19:54:57 +0100 Sebastian Dröge <sebastian@centricular.com>
* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
releasing 1.6.2
* gst-rtsp-server.doap:
Release 1.6.2
2015-11-11 14:58:33 +0100 Marcus Prebble <prebble@axis.com>
......
This is GStreamer 1.6.2
The GStreamer team is proud to announce the second bugfix release in the stable
1.6 release series of your favourite cross-platform multimedia framework!
This release only contains bugfixes and it is safe to update from 1.6.0 and
1.6.1. For a full list of bugfixes see Bugzilla:
https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&limit=0&list_id=83309&order=bug_id&product=GStreamer&resolution=FIXED&target_milestone=1.6.2
See http://gstreamer.freedesktop.org/releases/1.6/ for the latest version of this document.
Major bugfixes
- Crashes in gst-libav with sinks that did not provide a buffer pool
but supported video metadata were fixed. This affected d3dvideosink
and some 3rd party sinks. Also related fixes for crashes when a downstream
buffer pool failed allocation.
- Big GL performance improvement on iOS by a factor of 2 by using Apple's sync
extension.
- Deadlocks in the DirectSound elements on Windows, and the behaviour of its
mute property were fixed.
- The Direct3D video sink does not crash anymore when minimizing the window
- The library soname generation on Android >= 6.0 was fixed, which previously
caused GStreamer to fail to load there.
- File related elements have large-file (>2GB) support on Android now.
- gst-libav was updated to ffmpeg 2.8.3.
- Deserialization of custom events in the GDP depayloader was fixed.
- Missing OpenGL context initialization in the Qt/QML video sink was fixed in
certain situations.
- Interoperability with some broken RTSP servers using HTTP tunnel was
improved.
- Various compilation fixes for Windows.
- Various smaller memory leak and other fixes in different places.
- and many, many more:
https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&limit=0&list_id=83309&order=bug_id&product=GStreamer&resolution=FIXED&target_milestone=1.6.2
s is GStreamer 1.6.4
The GStreamer team is pleased to announce the fourth and likely last
bugfix release in the old stable 1.6 release series of your favourite
cross-platform multimedia framework!
This release only contains bugfixes and it should be safe to update from 1.6.x.
This release maintains API/ABI backwards compatibility with the
GStreamer 1.0, 1.2, 1.4 and 1.6 release series.
For details about the GStreamer 1.6 series and the latest version of this
document see the GStreamer 1.6 release page:
http://gstreamer.freedesktop.org/releases/1.6/
Bug fix summary:
- audio parsers: make sure to send tags before pushing the first buffer,
so all metadata is available at preroll. Fixes metadata collection in
mopidy with certain FLAC files.
- fix decoding glitches at the beginning of some mp3 streams when streaming
- multiqueue eos handling fixes
- tcpserversink/multisocketsink: fix 100% cpu usage on client disconnect
- video4linux: colorimetry and colorspace handling fixes
- udpsrc: add option to enable/disable multicast loopback ("loop" property)
- RTP JPEG: depayloader robustness fixes; payloader now accepts different
quant tables for the chroma components
- directsoundsink: fix some issues around muting/unmuting the sound
- dvdreadsrc: don't jump to wrong title when seeking back to 0 for titles != 1
- adaptivedemux: fix race on shutdown that could result in deadlocks
in hlsdemux/dashdemux, especially when stopped before playback started
- decklink: various robustness fixes in decklinkaudiosrc and decklinkvideosrc
- mpeg4parser: prevent assertion when scanning for sync code
- fbdevsink: fix crash caused by wrong bpp calculation
- tsdemux: fix hang in preroll caused by bogus timestamp/wraparound
handling in some corner cases
- tsdemux: fix accurating seeking
- h265parse: fix crash converting from hevc format to nal-aligned bytestream
- h264parse, h265parse: fix handling of downstream force-key-unit events
- g-i annotation fixes for bindings for gst_element_query_convert(),
gst_pad_get_current_caps(), and gst_pad_peer_query_caps()
- gst-libav: update internal libav copy to n2.8.6
- rtsp-server: report RECORD and ANNOUNCE as supported in the OPTIONS
- rtsp-server: prevent receival of looped back packets on Windows if a
multicast transport is used
- various minor memory leak fixes
- miscellaneous other fixes
- fix crashes on newer windows versions when GTypes are passed through
vararg functions as is done in souphttpsrc or during ges_init(). This
would manifest itself if the application was compiled with MSVC
and /DYNAMICBASE (address space layout randomization) was used.
- Bug list: https://bugzilla.gnome.org/buglist.cgi?product=GStreamer&target_milestone=1.6.4
Release notes for GStreamer RTSP Server Library 1.6.2
Release notes for GStreamer RTSP Server Library 1.6.4
The GStreamer team is proud to announce the second bugfix release in the stable
1.6 release series of your favourite cross-platform multimedia framework!
This release only contains bugfixes and it is safe to update from 1.6.0 and 1.6.1. For a
full list of bugfixes see Bugzilla.
See http://gstreamer.freedesktop.org/releases/1.6/
for the full release notes.
The GStreamer team is proud to announce a new bug-fix release
in the 1.x stable series of the GStreamer RTSP Server Library.
There were no bugs fixed in this release
Bugs fixed in this release
* 761226 : rtspmedia: Missing lock various functions (or the curious case of the duplicate unlocks)
* 761399 : rtspmedia: Missing lock in default_unsuspend, preroll_failed path
==== Download ====
You can find source releases of gst-rtsp-server in the download
directory: http://gstreamer.freedesktop.org/src/gst-rtsp-server/
directory: https://gstreamer.freedesktop.org/src/gst-rtsp-server/
The git repository and details how to clone it can be found at
http://cgit.freedesktop.org/gstreamer/gst-rtsp-server/
==== Homepage ====
The project's website is http://gstreamer.freedesktop.org/
The project's website is https://gstreamer.freedesktop.org/
==== Support and Bugs ====
......@@ -53,9 +46,12 @@ Interested developers of the core library, plugins, and applications should
subscribe to the gstreamer-devel list.
Applications
Contributors to this release
* Marcus Prebble
* Jan Schmidt
* Olivier Crête
* Sebastian Dröge
* Sebastian Rasmussen
* Srimanta Panda
* Steven Hoving
 
\ No newline at end of file
......@@ -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], [1.6.2],
AC_INIT([GStreamer RTSP Server Library], [1.6.4],
[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, 602, 0, 602)
AS_LIBTOOL(GST, 604, 0, 604)
dnl *** required versions of GStreamer stuff ***
GST_REQ=1.6.2
GSTPB_REQ=1.6.2
GSTPG_REQ=1.6.2
GSTPD_REQ=1.6.2
GST_REQ=1.6.4
GSTPB_REQ=1.6.4
GSTPG_REQ=1.6.4
GSTPD_REQ=1.6.4
dnl *** autotools stuff ****
......
......@@ -30,6 +30,16 @@ RTSP server library based on GStreamer
</GitRepository>
</repository>
<release>
<Version>
<revision>1.6.4</revision>
<branch>1.6</branch>
<name></name>
<created>2016-04-14</created>
<file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.6.4.tar.xz" />
</Version>
</release>
<release>
<Version>
<revision>1.6.2</revision>
......
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