Release 1.7.1

1 files changed, 235 insertions, 2 deletions

diff --git a/ChangeLog b/ChangeLog
index 62ce38f..a931568 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,9 +1,242 @@
+=== 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 ===