diff options
author | Sebastian Dröge <sebastian@centricular.com> | 2015-12-24 14:54:06 +0100 |
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committer | Sebastian Dröge <sebastian@centricular.com> | 2015-12-24 14:54:06 +0100 |
commit | 7374976722852c93e59c3cee2a562f947cb7f310 (patch) | |
tree | a4de825f383c2768537ad345073ad6a9b9d5af4d /ChangeLog | |
parent | c934fdaf3b63ed156ef4dbc73afaf59c4e30a6fa (diff) |
Release 1.7.1gst-rtsp-server-1.7.1
Diffstat (limited to 'ChangeLog')
-rw-r--r-- | ChangeLog | 237 |
1 files changed, 235 insertions, 2 deletions
@@ -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 === |