+=== release 1.7.1 ===
+
+2015-12-24 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ 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 Sebastian Dröge <slomo@coaxion.net>
+2015-09-25 23:32:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.6.0
+ * gst-rtsp-server.doap:
+ Release 1.6.0
=== release 1.5.91 ===
-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
-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 ====
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
\ No newline at end of file
projects / platform / upstream / gstreamer.git / commitdiff