diff options
| -rw-r--r-- | ChangeLog | 3957 | ||||
| -rw-r--r-- | NEWS | 20 | ||||
| -rw-r--r-- | RELEASE | 85 | ||||
| -rw-r--r-- | configure.ac | 4 | ||||
| -rw-r--r-- | gst-rtsp-server.doap | 10 |
5 files changed, 4073 insertions, 3 deletions
@@ -0,0 +1,3957 @@ +=== release 1.2.3 === + +2014-02-25 Tim-Philipp Müller <tim@centricular.com> + + * configure.ac: + releasing 1.2.3 + +2014-02-25 20:29:14 +0000 Tim-Philipp Müller <tim@centricular.com> + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.2.3 + +2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com> + + * examples/Makefile.am: + examples: use LDADD for libs instead of LDFLAGS + +2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com> + + * configure.ac: + configure: make sure releases are in .doap file + +2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com> + + * examples/test-cgroups.c: + examples: test-cgroups: don't put code with side effects into g_assert() + The g_assert() might get compiled out with the right + compiler/preprocessor flags. + +2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com> + + * examples/.gitignore: + examples: add cgroup test binary to .gitignore + +2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com> + + * examples/test-cgroups.c: + examples: fix cgroup test build + Fixes build failure caused by compiler warning: + test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes] + +2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com> + + * .gitignore: + .gitignore: ignore temp files created in the course of 'make check' + +2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: don't loose frames handling new PLAY request + If client supplied a range check if the range specifies the start point. + If not, then do an accurate seek to the current position. If a start + point was specified do do a key unit seek to make sure the streaming + starts with decodeable frames. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611 + +2014-02-17 10:43:05 +0100 Göran Jönsson <goranjn@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: set ttl-mc before adding the socket + Set ttl-mc before adding the socket. Otherwise the value ttl-mc will + never be set on socket. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531 + +2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com> + + * gst/rtsp-server/rtsp-media.c: + media: stop thread if media is already prepared + in gst_rtsp_media_prepare() the thread is not used if media is already + prepared (e.g. media shared) so we want to stop the thread. otherwise, a + leak occurs. + https://bugzilla.gnome.org/show_bug.cgi?id=724182 + +=== release 1.1.90 === + +2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com> + + * Makefile.am: + build: Ship gst-rtsp-server.doap file + +2014-02-09 10:51:23 +0100 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + configure: Set version to 1.1.90 for pre-release and require GStreamer 1.2.3 or newer + +2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com> + + * tests/check/gst/rtspserver.c: + tests: Fix another compiler warning with gcc + +2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/client.c: + rtsp-server: Fix lots of compiler warnings with clang + +2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + * gst-rtsp-server.doap: + * tests/Makefile.am: + configure: Synchronise with the configure scripts of the other modules + +2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + configure: Update version to 1.3.0.1 and require GStreamer 1.3.0 + +2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + Revert "rtsp-server: support build against last stable release" + This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f. + Let us require 1.2.3 now, which is going to be released in a few + minutes. + +2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-stream-transport.c: + session: improve RTP-Info + Ignore streams that can't generate RTP-Info instead of failing. + Don't return the empty string when all streams are unconfigured but + return NULL so that we don't generate and empty RTP-Info header. + Improve docs a little. + +2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com> + + * gst/rtsp-server/rtsp-session-media.c: + Don't free rtpinfo GString when it is NULL + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554 + +2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-media.c: + media: only set keyframe flag when modifying start + Only set the keyframe flag when we modify the start position. The + keyframe flag should probably be ignored when no change is requested but + until we can claim this is all documented properly and all demuxer + implement this, avoid setting the flag. + See also https://bugzilla.gnome.org/show_bug.cgi?id=723075 + +2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-thread-pool.c: + thread-pool: Unref source after mainloop has quit to avoid races in GLib + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741 + +2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: handle NULL seqnum and rtptime arguments + +2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-thread-pool.c: + * tests/check/gst/threadpool.c: + thread-pool: Unref reused threads in gst_rtsp_thread_stop() + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519 + +2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: add fallback for missing stats property + Use a fallback when the payloader does not have a stats property + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554 + +2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com> + + * common: + Automatic update of common submodule + From f7bc1c3 to 1a07da9 + +2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: don't leak stats structure + Don't leak the stats structure and deal with NULL stats. + +2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: Get rtpinfo properties atomically from payloader + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844 + +2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-media.c: + media: refactor state change functions and signals + Make functions to set the target state and the pipeline state and emit + the signals from those functions. + +2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add signal to notify of pending state changes + +2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + rtsp-server: support build against last stable release + Until 1.2.3 is out with the new get_type function and we + can require that. + +2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: fix compilation + +2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add property to configure profiles + +2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-client.c: + client: let stream check supported transport + Delegate the check if a transport is allowed to the stream. + See https://bugzilla.gnome.org/show_bug.cgi?id=720696 + +2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add method to check supported transport + Add a method to check if a transport is supported + +2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com> + + * configure.ac: + configure.ac: Only check for gstreamer-check, not check + We include check in gstreamer-check since quite some time now. + +2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: return clock-rate from get_rtpinfo + And use it to correct the rtptime to the requested start-time. + See https://bugzilla.gnome.org/show_bug.cgi?id=712198 + +2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + session-media: calculate start-time + +2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: also return the running-time + Return the running-time in the rtpinfo as well. + +2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-media.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + session-media: let the session-media make the RTPInfo + Add method to create the RTPInfo for a stream-transport. + Add method to create the RTPInfo for all stream-transports in a + session-media. + Use the session-media RTPInfo code in client. This allows us to refactor + another method to link the TCP callbacks. + +2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com> + + mount-points: sort sequence before g_sequence_lookup + * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory): + sort sequence if dirty, otherwise lookup will fail. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855 + +2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com> + + * configure.ac: + configure: rename package from gst-rtsp to gst-rtsp-server + To match git module name and avoid confusion with the + rtsp lib in gst-plugins-base and rtsp plugin in -good. + +2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com> + + * configure.ac: + configure: bump core/base/good requirement to 1.2.0 + Bump to released stable version and make implicit + requirements explicit. + +2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com> + + * autogen.sh: + * common: + * configure.ac: + Fix broken gettext setup which is not used anyway + +2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com> + + * common: + Automatic update of common submodule + From dbedaa0 to d48bed3 + +2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add setup_sdp vmethod + gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public + gst_rtsp_media_setup_sdp. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155 + +2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Check return value of sscanf + streamid is only valid if sscanf matched something. + +2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Fix iteration + Wouldn't even enter the code block otherwise (i++ was used as the check + and not the postfix). + +2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: add vmethod to configure media and streams + Implement a vmethod that can be used to configure the media and the + streams based on the current context. Handle the blocksize handling in + the default handler. + See https://bugzilla.gnome.org/show_bug.cgi?id=720667 + +2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com> + + * .gitignore: + Make git ignore more unit test binaries + +2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/rtsp-address-pool.h: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-context.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-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 padding to many public structures + Not mini objects though, since they are not subclassable + anyway, nor kept on the stack or inlined in a structure. + +2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com> + + media: add new create_rtpbin vmethod + * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod. + https://bugzilla.gnome.org/show_bug.cgi?id=719734 + +2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com> + + * tests/check/gst/media.c: + tests: fix memory leak, free test's thread pool + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733 + +2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-stream-transport.c: + stream-transport: free url in finalize + +2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-media.c: + media: also do state change in suspended state + +2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + media: also handle prepare and range in suspended state + When we are suspended, we are already prepared. + We can get the range in the suspended state. + +2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com> + + * tests/check/Makefile.am: + * tests/check/gst/sessionmedia.c: + check: add test for uri in setup + Added unit tests for the new functionality in GstRTSPStreamTransport. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168 + +2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-client.c: + client: store setup uri and use in PLAY response + Store the uri used when doing the setup and use that in the PLAY + response. + fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168 + +2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + stream-transport: add method to get/set url + +2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-client.c: + client: suspend after SDP and unsuspend before PLAYING + Based on patches by Ognyan Tonchev <ognyan@axis.com> + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257 + +2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session.c: + * tests/check/gst/media.c: + * tests/check/gst/mediafactory.c: + media: add suspend modes + Add support for different suspend modes. The stream is suspended right after + producing the SDP and after PAUSE. Different suspend modes are available that + affect the state of the pipeline. NONE leaves the pipeline state unchanged and + is the current and old behaviour, PAUSE will set the pipeline to the PAUSED + state and RESET will bring the pipeline to the NULL state. + A stream is also unsuspended when it goes back to PLAYING, for RESET streams, + this means that the pipeline needs to be prerolled again. + Base on patches by Ognyan Tonchev <ognyan@axis.com> + See https://bugzilla.gnome.org/show_bug.cgi?id=711257 + +2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-media.c: + media: start live streams in blocked state + Start live streams in the blocked state and make them preroll using the + messages. This ensure that no data is played by the sink until we explicitly + unblock the stream right before going to PLAYING. + See https://bugzilla.gnome.org/show_bug.cgi?id=711257 + +2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-media.c: + media: refactor starting and waiting for preroll + Based on patches from Ognyan Tonchev <ognyan@axis.com> + See https://bugzilla.gnome.org/show_bug.cgi?id=711257 + +2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add API to block streams + Add an API to block on the streams and make it post a message. + Based on patch by Ognyan Tonchev <ognyan@axis.com> + See https://bugzilla.gnome.org/show_bug.cgi?id=711257 + +2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com> + + * docs/libs/Makefile.am: + docs: Specify the override file + Even if it's empty (for now) it avoids make distcheck complaining + +2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-media.c: + media: move default implementations to where they are used + +2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-media.c: + media: take the right lock in gst_rtsp_media_set_pipeline_state() + We need to take the state_lock when calling this method. + +2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-media.c: + media: handle add-added on non-bins too + Handle dynamic payloaders that are not bins, as used in the unit-test. + +2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + rtsp-media/-factory: Fix request pad name comments + These must be escaped for gtk-doc to parse the comments without warnings. + +2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com> + + rtsp-media: remove transports if media is in error status + * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are + trying to change to GST_STATE_NULL and media is in error status, we + remove all transports. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776 + +2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: use element metadata to find payloader + Use the element metadata to find the payloader instead of checking + for the base class. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396 + +2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com> + + rtsp-stream: add getter for payload type + * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt. + * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader + element and create the stream with this one instead of the dynpay%d + element. + https://bugzilla.gnome.org/show_bug.cgi?id=712396 + +2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-context.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-token.c: + rtsp-*: Refer to NULL as a constant in comments + Plus one typo fix. + https://bugzilla.gnome.org/show_bug.cgi?id=714988 + +2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + rtsp-*: Fix type name typos in comments + * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken + * rtsp-auth: Refer to part of constant name as text + * rtsp-auth/-permissions/-token: Refer to Permissions not Permission + * rtsp-session-media: Fix GstRTSPSessionMedia typo + * rtsp-stream: Fix typo when refering to GstBin + https://bugzilla.gnome.org/show_bug.cgi?id=714988 + +2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * docs/README: + * docs/libs/gst-rtsp-server-docs.sgml: + * docs/libs/gst-rtsp-server-sections.txt: + docs: Improve documentation + * Include annotation-glossary to quiet gtk-doc + * Rename remaining ClientState -> Context + * Rename object hierarchy file + * Remove stale chapter references + * Add missing function and object references + * Include missing GstRTSPAddressPoolResult + https://bugzilla.gnome.org/show_bug.cgi?id=714988 + +2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-stream.c: + rtsp-server: sprinkle some allow-none annotations for g-i + +2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add method to filter transports + Add a method to safely iterate and collect the stream transports + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664 + +2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + rtsp: allow NULL func in filters + Passing a null function make the filters return a list of + refcounted objects. + +2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com> + + * gst/rtsp-server/rtsp-address-pool.c: + * tests/check/gst/addresspool.c: + address-pool: fix address increment + Use a guint instead of guint8 to increment the address. It's still not + completely correct because a guint might not be able to hold the complete + address range, but that's an enhacement for later. + Add unit test to test improved behaviour. + https://bugzilla.gnome.org/show_bug.cgi?id=708237 + +2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * tests/check/gst/client.c: + client: allow absolute path in requests + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689 + +2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: make make_path_from_uri a vmethod + +2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/Makefile.am: + * tests/check/gst/stream.c: + stream: Add functions to get rtp and rtcp sockets + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100 + +2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com> + + * gst/rtsp-server/rtsp-context.c: + * gst/rtsp-server/rtsp-context.h: + context: defing a GType for the context + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018 + +2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org> + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-context.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-stream.c: + Fixed several GIR warnings + +2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com> + + * gst/rtsp-server/rtsp-auth.c: + auth: small typos + +2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com> + + * tests/check/Makefile.am: + * tests/check/gst/token.c: + tests: Add unit tests for token + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520 + +2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com> + + * gst/rtsp-server/rtsp-token.c: + token: Validate args for gst_rtsp_token_is_allowed + See https://bugzilla.gnome.org/show_bug.cgi?id=710520 + +2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com> + + * gst/rtsp-server/rtsp-token.c: + token: Fix bug when creating empty token + We always want to have a valid GstStructure in the token. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520 + +2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com> + + * gst/rtsp-server/rtsp-thread-pool.c: + thread-pool: avoid race in shutdown + If we call g_main_loop_quit before the thread has entered g_main_loop_run, we + don't actually stop the mainloop ever. Solve this race by adding an idle source + to the mainloop that calls the _quit. This way we immediately exit the mainloop + if quit was called before we started it. + +2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com> + + * tests/check/Makefile.am: + * tests/check/gst/permissions.c: + tests: Add unit tests for permissions + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202 + +2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com> + + * tests/check/gst/mediafactory.c: + tests: Test mediafactory permissions + See https://bugzilla.gnome.org/show_bug.cgi?id=710202 + +2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com> + + * gst/rtsp-server/rtsp-permissions.c: + permissions: Fix refcounting when adding/removing roles + Previously a role that was removed was unreffed twice, and when + replacing an existing role the replaced role was freed while still being + referenced. Both bugs are now fixed. + See https://bugzilla.gnome.org/show_bug.cgi?id=710202 + +2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com> + + * tests/check/gst/media.c: + * tests/check/gst/mediafactory.c: + * tests/check/gst/rtspserver.c: + tests: Check gst_rtsp_url_parse return value + See https://bugzilla.gnome.org/show_bug.cgi?id=710202 + +2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com> + + * common: + Automatic update of common submodule + From 865aa20 to dbedaa0 + +2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-server.c: + rtsp-server: Fix socket leak + https://bugzilla.gnome.org/show_bug.cgi?id=710088 + +2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com> + + * gst/rtsp-server/rtsp-session-pool.c: + rtsp-session-pool: Make sure session IDs are properly URI-escaped + https://bugzilla.gnome.org/show_bug.cgi?id=643812 + +2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com> + + * examples/.gitignore: + * examples/test-video.c: + examples: fix compilation when WITH_AUTH is defined + https://bugzilla.gnome.org/show_bug.cgi?id=710228 + +2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com> + + * .gitignore: + gitignore: Add new test binary + +2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com> + + * tests/check/Makefile.am: + * tests/check/gst/threadpool.c: + thread-pool: Add unit test for the thread pools + https://bugzilla.gnome.org/show_bug.cgi?id=710228 + +2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-thread-pool.c: + thread-pool: Fix thread leak when reusing threads + https://bugzilla.gnome.org/show_bug.cgi?id=709730 + +2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-server.c: + * tests/check/gst/rtspserver.c: + tests: fixed racy behavior in rtspserver tests + https://bugzilla.gnome.org/show_bug.cgi?id=710078 + +2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com> + + * tests/check/gst/addresspool.c: + tests: Improve address pool unit tests + Add a range with mixed IPV4 and IPV6 addresses to pool. + Get an IPV4 address from an IPV6-only pool. + Get an IPV6 address from an IPV4-only pool. + Reserve a IPV6 address from an IPV4-only pool. + Check for unicast addresses in multicast-only pool. + Check for unicast addresses in uni-/multicast-mixed pool. + https://bugzilla.gnome.org/show_bug.cgi?id=710128 + +2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: append query string in PAUSE/PLAY/TEARDOWN as well + +2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com> + + * gst/rtsp-server/rtsp-client.c: + client: Add query to control path + If the SETUP url contains a query it must be appended to the control + path so that it matches any already created stream in the media. The + query will also be appended to the session media path. + +2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: remove old line + +2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: Correct control comparison + https://bugzilla.gnome.org/show_bug.cgi?id=709176 + +2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: Check dynamically if the pipeline supports seeking + We should not depend on whether or not the pipeline state change + returned NO_PREROLL or not. A media could dynamically change its + element and switch from seekable to non seekable so it's best to test + the seekable nature of the pipeline dynamically when we try to do a seek. + +2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: Return FALSE if seeking is not supported + +2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: don't seek accurate by default + Accurate seeking is perhaps a little overkill in the most common situation and + causes some formats (mp3) over slow media to seek extremely slowly. + +2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com> + + * tests/check/gst/rtspserver.c: + tests: fix unit test + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742 + +2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com> + + * gst/rtsp-server/rtsp-client.c: + client: Reply 400 if media cannot be constructed + Reply 400 Bad Request instead of 503 Service Unavailable if media + cannot be constructed in SETUP. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821 + +2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com> + + * gst/rtsp-server/rtsp-client.c: + client: Send setup reply once only + If find_media() failed in handle_setup_request() two replies was sent. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819 + +2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net> + + * common: + Automatic update of common submodule + From 6b03ba7 to 865aa20 + +2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com> + + * gst/rtsp-server/rtsp-server.c: + server: Emit client-connected signal earlier + Emit client-connected before the client ref is given to a GSource, + otherwise client-connected can be emitted after the client object has + been freed. + +2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com> + + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-address-pool.h: + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/addresspool.c: + addresspool: return reason of failure + Let gst_rtsp_address_pool_reserve_address() return the reason why + the address could not be reserved. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229 + +2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com> + + * autogen.sh: + autogen.sh: Sync behaviour with other GStreamer modules + Allows building from outside of tree amongst other things + +2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com> + + * common: + Automatic update of common submodule + From b613661 to 6b03ba7 + +2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net> + + * common: + Automatic update of common submodule + From 74a6857 to b613661 + +2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net> + + * common: + Automatic update of common submodule + From 01a7a46 to 74a6857 + +2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com> + + * gst/rtsp-server/rtsp-client.c: + client: Do not read beyond end of path string + If the setup was done without a control url, make sure we don't try to read the + non-existing control string and crash. + +2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: Fix RTPInfo header + Refactor the method to make the content_base. + Use the content-base and the control url to construct the RTPInfo + url. + +2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: map url to path only in describe + Only map the request url to a path in the DESCRIBE method. The SDP then + contains the base and control urls that should be used to SETUP/PAUSE/ + PLAY/TEARDOWN the media. + +2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + Revert "client: map URL to path in requests" + This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d. + This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then + contains the base and control urls which are used in the SETUP, PLAY, + PAUSE and TEARDOWN requests. + +2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: map URL to path in requests + +2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-mount-points.h: + mount-points: make vmethod to make path from uri + Make a vmethod to transform an url into a path. The path is then used to lookup + the factory. This makes it possible to also use other bits of the url, such as + the query parameters, to locate the factory. + +2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-thread-pool.c: + * gst/rtsp-server/rtsp-thread-pool.h: + thread-pool: Add cleanup to wait for the threadpool to finish + Also fix race condition if two threads are asking for the first + thread from the thread pool at once. This would case two internal + GThreadPools to be created. + https://bugzilla.gnome.org/show_bug.cgi?id=707753 + +2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * tests/check/gst/client.c: + client: free threadpool + https://bugzilla.gnome.org/show_bug.cgi?id=707638 + +2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com> + + * tests/check/gst/mountpoints.c: + mountpoints tests: unref matched factories + https://bugzilla.gnome.org/show_bug.cgi?id=707638 + +2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com> + + * tests/check/gst/media.c: + media tests: unref thread pool and caps + https://bugzilla.gnome.org/show_bug.cgi?id=707638 + +2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + auth, media, media-factory: unref permissions + https://bugzilla.gnome.org/show_bug.cgi?id=707638 + +2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/Makefile.am: + Makefile: add rule for appsrc example + +2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-appsrc.c: + tests: add appsrc example + Add an example on how to use appsrc to feed the server pipeline with data. + +2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: remove query part from content-base string + Make sure that after the control url has been resolved, it's + not a part of the query-string. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568 + +2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: don't check url in response + There is no url or method in the response to check + +2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + Add handle-response signal for when we receive a GET_PARAMETER response + +2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + Fix gst_rtsp_server_client_filter, using wrong variable type + +2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net> + + * gst/rtsp-server/rtsp-media-factory-uri.c: + rtsp-media-factory-uri: check AAC properly for whether it's parsed or not + For AAC we need to check for framed=true instead of parsed=true. + https://bugzilla.gnome.org/show_bug.cgi?id=701384 + +2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + stream: optimize pipeline for protocols + When TCP is not an allowed protocol for the stream, avoid creating the + appsrc/appsink/queue and tee elements. + +2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: set protocols on streams + +2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: use protocols supported by stream + +2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + media-factory: allow all protocols + +2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: configure protocols in new streams + +2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add protocols property + +2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: send state in "new-state" signal + https://bugzilla.gnome.org/show_bug.cgi?id=705110 + +2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com> + + * configure.ac: + build: add subdir-objects to AM_INIT_AUTOMAKE + Fixes warnings with automake 1.14 + https://bugzilla.gnome.org/show_bug.cgi?id=705350 + +2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: add method to iterate clients of server + +2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + Add vmethod for rtsp-media subclass to access rtpbin + +2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.h: + small documentation fix + +2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + Do not take range header if range is invalid + +2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-media.c: + media: add docs for new method + +2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + Add API to rtsp-media set the pipeline's state + +2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + Update current position/duration when gst_rtsp_media_get_range_string is called + +2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-cgroups.c: + tests: add some more docs + +2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-cgroups.c: + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-context.c: + * gst/rtsp-server/rtsp-context.h: + * gst/rtsp-server/rtsp-params.c: + * gst/rtsp-server/rtsp-params.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-thread-pool.c: + * gst/rtsp-server/rtsp-thread-pool.h: + * tests/check/gst/client.c: + ClientState -> Context + Rename the clientstate to context and put the code in a separate file. + +2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + auth: add support for default token + The default token is used when the user is not authenticated and can be used to + give minimal permissions. + +2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + * gst/rtsp-server/rtsp-auth.c: + auth: use defines when possible + +2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-address-pool.c: + address-pool: improve docs + +2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-permissions.c: + permissions: add the role to the copy + +2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-permissions.c: + permissions: Also copy the roles + +2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-permissions.c: + permissions: Make it build + +2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-address-pool.h: + docs: small fixes + +2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/client.c: + docs: improve docs + +2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-address-pool.h: + * tests/check/gst/addresspool.c: + * tests/check/gst/rtspserver.c: + address-pool: cleanups + Remove redundant method, improve docs. + +2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-permissions.c: + * gst/rtsp-server/rtsp-permissions.h: + * gst/rtsp-server/rtsp-token.c: + docs: improve docs + +2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-permissions.c: + permissions: implement _remove_role + +2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-permissions.c: + permissions: update docs + +2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/gst/client.c: + tests: simplify tests + Client settings are now disabled by default so we don't need an auth + module to disable them. + +2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-auth.c: + auth: add default authorizations + When no auth module is specified, use our table of defaults to look up the + default value of the check instead of always allowing everything. This was + we can disallow client settings by default. + +2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/README: + README: update readme + +2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-thread-pool.c: + * gst/rtsp-server/rtsp-thread-pool.h: + thread-pool: add more docs + +2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-thread-pool.c: + * gst/rtsp-server/rtsp-thread-pool.h: + thread-pool: fix race in thread reuse + If we try to reuse a thread right after we made it stop, we end up using a + stopped thread. Catch this case and only reuse threads that are not stopping. + +2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + server: add small debug + +2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/gst/client.c: + client: fix test + Add some permissions to media so we can use the auth and enable + client settings. + +2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: support pushed context in handle_request + If we already have a pushed state, reuse it and add our own things. This makes + it easier to write tests. + +2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-auth.c: + auth: don't auth on methods + Don't authorize on methods anymore but on the resources that we + try to access, this is more flexible. + Move the authorization checks to where they are needed and let the + check return the response on error. + +2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-mount-points.c: + mount-points: add some debug + +2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/gst/client.c: + tests: almost fix test + +2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + auth: let the auth module check client_settings + Let the auth module decide if client settings are allowed for the + current client. + +2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-token.c: + * gst/rtsp-server/rtsp-token.h: + token: add method to check boolean permission + +2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + * examples/test-cgroups.c: + * gst/rtsp-server/rtsp-token.c: + * gst/rtsp-server/rtsp-token.h: + token: simplify token constructor + Use variable arguments to make easier API. + +2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + * examples/test-cgroups.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: add convenience API for factory + +2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + * examples/test-cgroups.c: + * gst/rtsp-server/rtsp-permissions.c: + * gst/rtsp-server/rtsp-permissions.h: + permissions: simplify API a little + Avoid passing GstStructure in the add_role method, use varargs instead + to construct the structure behind the scenes. We can then also use the + structure name as the role and simplify some more logic. + +2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-auth.c: + auth: fix typo + +2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + auth: handle unauthorized response + Move handling of the unauthorized response to the auth module, it can add + the appropriate headers to request authorization for the required method + much better than the client. + +2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: allow for sending any message, not only requests + Change the _send_request() method to _send_message() so that we + can both send requests and replies. + +2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-server.h: + docs: fix docs + +2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-video.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + auth: move TLS handling to auth module + Remove the TLS settings on the server and move it to the auth module because + that is where security related bits go. + +2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: add state push/pop + +2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: add connection to state + +2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-mount-points.c: + mount-points: fix debug + +2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/gst/media.c: + tests: fix media test + +2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-thread-pool.c: + thread-pool: we don't require a state + +2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + server: let context ref the server + So that we don't risk losing the server object early anc crash. + +2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/gst/client.c: + tests: fix client test + +2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/README: + * docs/libs/gst-rtsp-server-docs.sgml: + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-params.c: + * gst/rtsp-server/rtsp-permissions.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-thread-pool.c: + * gst/rtsp-server/rtsp-token.c: + docs: improve docs + +2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session-pool.h: + session-pool: make vmethod to create a session + Make a vmethod to create a sessions so that subclasses can create + custom session objects + +2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-mount-points.h: + * gst/rtsp-server/rtsp-session-pool.h: + * gst/rtsp-server/rtsp-stream.h: + docs: more updates + +2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-docs.sgml: + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-address-pool.h: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-permissions.c: + * gst/rtsp-server/rtsp-permissions.h: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session-media.c: + * 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.c: + * gst/rtsp-server/rtsp-thread-pool.h: + docs: update docs + +2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * configure.ac: + * examples/Makefile.am: + configure: compile cgroup example conditionally + Only compile the cgroup example when we have libcgroup + +2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * configure.ac: + * examples/Makefile.am: + * examples/test-cgroups.c: + examples: add cgroups example + +2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/gst/rtspserver.c: + tests: fix compilation + +2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-thread-pool.c: + thread-pool: fix vmethod invocation + +2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-thread-pool.c: + * gst/rtsp-server/rtsp-thread-pool.h: + thread-pool: store thread type in thread + +2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: pass thread from pool to media _prepare + Get a thread from the configured threadpool and pass it to the prepare method of + the media. + +2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: Accept a thread in _prepare + Remove out own threadpool handling and use the provided thread and + maincontext for the bus messages and the state changes. + +2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + server: configure client thread pool + +2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: add method to configure thread pool + +2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: use thread pool + Use the thread pool instead of doing our own thing. + +2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-thread-pool.c: + * gst/rtsp-server/rtsp-thread-pool.h: + thread-pool: add object to manage threads + Add an object to manage the client and media threads. + +2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-auth.c: + auth: debug authorization check + +2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: start media pipeline in context + Start the media pipeline in the provided context (or our default one + when NULL). This makes sure that we run the bus thread in this context and that + all media threads are children of this context. + +2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-factory.c: + factory: pass permissions to media by default + +2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + test: add permissions to auth test + Ass some permissions to the media factory in the test. + +2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + auth: simplify auth checks + Remove client from methods, it's now in the state + Perform the check specified by the string, use the information from the + thread local context. + +2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: add state to current thread + Add the client to the ClientState object. + Place the ClientState on the current thread. + +2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: make it possible to set permissions + Make it possible to set permissions on media and media factory objects + +2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-permissions.c: + * gst/rtsp-server/rtsp-permissions.h: + permissions: add permissions object + Add a mini object to store permissions based on a role. + +2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + auth: add auth checks + Add an enum with auth checks and implement the checks in the auth object. + Perform the checks from the client. + +2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.h: + auth: use the token after authentication + After we authenticated a user, keep the Token around in the state. + +2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * tests/check/gst/media.c: + media: add optional context for bus messages + Add an optional mainloop to _prepare that will handle the bus messages instead + of always using the shared mainloop. + +2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-token.c: + * gst/rtsp-server/rtsp-token.h: + token: add authorization token + Add a simply miniobject that contains the authorizations. The object contains a + GstStructure that hold all authorization fields. When a user is authenticated, + the auth module will create a Token for the user. The token is then used to + check what operations the user is allowed to do and various other configuration + values. + +2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + auth: remove auth from media and factory + Remove the auth object from media and factory. We want to have the RTSPClient + authenticate and authorize resources, there is no need to place another auth + manager on the media/factory. + +2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.h: + auth: add support for multiple basic auth tokens + Make it possible to add multiple basic authorisation tokens to one authorization + object. Associate with each token an authorization group that will define what + capabilities are allowed. + +2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: error out on non-aggregate control + We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN. + +2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: rework setup request a little + Cache the media in DESCRIBE based on the longest matching path with the uri + that we can find in the mount points. + Rework the setup request a little to get the media from the session or from + the longest matching path, this way we can derive the control string as + everything after the path instead of hardcoding it. + Find the stream based on the control string and only open a session when all + this can be done. + +2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add method to find a stream by control url + +2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add method to check control url of stream + +2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-media.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + session: use path matching for session media + Use a path string instead of a uri to lookup session media in the sessions. Also + use path matching to find the largest possible path that matches. + +2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-mount-points.h: + * tests/check/gst/mountpoints.c: + mount-points: remove useless vmethod + Making lookups in the mount points should not be done with a URL, if there is a + mapping to be done from URL to mount points, we'll need to do it somewhere + else. + +2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-mount-points.h: + * tests/check/gst/mountpoints.c: + mount-points: improve mount point searching + Use a GSequence to keep track of the mount points. + Match a URL to the longest matching registered mount point. This should be the + URL to perform aggreagate control and the remainder is the stream specific + control part. + Add some unit tests for this. + +2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org> + + * gst/rtsp-server/Makefile.am: + rtsp-server: Allow building of static library + +2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/gst/mediafactory.c: + tests: fix compilation + +2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-sdp.c: + sdp: get control string from stream + Use the control string as configured in the stream. + +2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add methods and property to set control string + +2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: cleanups + Rename variables for clarity + Keep media in state when we can + +2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add more support for IPv6 + Rename _get_address to _get_multicast_address in GstRTSPStream to + make it clear that this function only deals with multicast. + Make it possible to have both an IPv4 and IPv6 multicast address on + a stream. Give the client an IPv4 or IPv6 address depending on the + address it used to connect to the server. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002 + +2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: fix comment + +2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + stream: handle failed port allocation + Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we + can't allocate any family at all. Also keep track of what port families we + allocated. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175 + +2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + stream: improve docs + +2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream-transport.c: + stream-transport: remove old if 0 block + +2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com> + + * tests/check/gst/client.c: + tests: fix tests + gst_rtsp_client_get_uri() has been removed + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173 + +2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: add method to filter managed sessions + Add a method to filter the sessions managed by this client connection. + See https://bugzilla.gnome.org/show_bug.cgi?id=703016 + +2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: remove _get_uri() method + Remove the get_uri() method on the client. A client has no uri, the uri + property is an internal property to manage the last cached media for + the client. + +2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: fix typo + +2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Do not leak the query in default_query_stop + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120 + +2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: don't unlock when conversion fails + Don't unlock the state lock when conversion fails because it was not locked. + +2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + Add query_position and query_stop vmethods to rtsp-media + +2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + Fix typo in property install for rtsp-media's time-provider + +2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: clean some variables + Clean some variables and add some guards to _send_request() + +2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + Add gst_rtsp_client_send_request API + This makes it possible to send arbitrary messages to a client, such as + SET_PARAMETER or GET_PARAMETER + +2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add _get_element() method + Add method to get the element used when creating the media. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008 + +2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: fix docs + +2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: allow access to the rtp session + https://bugzilla.gnome.org/show_bug.cgi?id=703004 + +2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + dscp qos support in gst-rtsp-stream + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645 + +2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/gst/rtspserver.c: + tests: fix test + Actually do what the comment says. Also keep the old code around, not sure what + should happen when you get a 454 from a TEARDOWN, does it close the connection? + it currently doesn't. + +2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: also watch newly created session + When we newly created a session, start watching it immediately instead of + on the next request. + +2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com> + + * tests/check/gst/client.c: + tests: add unit test for new-session + See https://bugzilla.gnome.org/show_bug.cgi?id=701587 + +2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: emit new-session when new session is created + Only emit new-session when we created a new session for a client, not when a + client picked up a previous session. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587 + +2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com> + + * gst/rtsp-server/rtsp-client.c: + client: handle asterisk as path in requests + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266 + +2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: handle segment query format mismatch + It's possible that the segment query returns with a different format than what + we asked for, handle this case also. + +2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-media.c: + media: use segment stop in collect_media_stats + Use segment stop instead of duration as range end point. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185 + +2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-media.c: + * tests/check/gst/media.c: + rtsp-media: Do not leak the element in take_pipeline + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470 + +2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + rtsp-client: Make configure_client_transport virtual + This patch makes configure_client_transport virtual. The functionality is + needed to handle some weird clients sending multicast transport settings as url + options. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173 + +2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + rtsp-client: Make param_set and param_get virtual + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072 + +2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: convert_range replaces get_range_times + get_range_times worked for handling UTC ranges for seeks, but we also + need to convert back from NPT to the requested unit in + get_range_string. convert_range is now used for both. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084 + +2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-sdp.h: + sdp: cleanup sdp info + We don't need to pass the proto, we can more easily check a boolean. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063 + +2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com> + + * gst/rtsp-server/rtsp-sdp.c: + use 0.0.0.0 or :: for c= line instead of server address + +2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com> + + * gst/rtsp-server/rtsp-client.c: + use local address, not remote, in SDP + See https://bugzilla.gnome.org/show_bug.cgi?id=702063 + +2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * common: + Automatic update of common submodule + From 098c0d7 to 01a7a46 + +2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: possibility to override range time conversion + Make it possible to override the conversion from GstRTSPTimeRange to + GstClockTimes, that is done before seeking on the media + pipeline. Overriding can be useful for UTC ranges, where the default + conversion gives nanoseconds since 1900. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191 + +2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + rtsp-server: Expose the use_client_settings API + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935 + +2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + rtspstream: handle both ipv4 and ipv6 clients + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129 + +2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-sdp.c: + Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute" + This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97. + We already have a way to place extra attributes in the SDP by using a string + property with prefix x- or a- in the caps. + +2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-sdp.c: + Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute" + This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494. + We already have a way to place extra attributes in the SDP, just make a string + property in the payloader with a- or x- prefix. + +2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-sdp.c: + rtsp: place a- and x- properties as attributes + application/x-rtp has properties with a- and x- prefixes that should be + placed as attributes in the SDP for the media instead of being added to the + fmtp. + +2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/Makefile.am: + * examples/test-video.c: + example: add TLS example + +2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: add support for TLS + Add methods to set and get a TLS certificate. + Add vmethod to configure a new connection. By default, configure the TLS + certificate in a new connection if needed. + +2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: remove accept_client vmethod + This vmethod is not very useful so remove it. + +2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + server: don't crash on NULL GError + +2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-session-pool.c: + rtsp-session-pool: corrected session timeout detection + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253 + +2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: improve debug + +2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-server.c: + server: refactor connection setup + Let the server accept the socket connection and construct a GstRTSPConnection + from it. Remove the code from the client and let the client only deal with + a fully configure GstRTSPConnection object. + We will need this later when the server will configure the connection for + TLS. + +2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + stream: keep the transport object alive + Keep the transport object alive while we have it as qdata on the + source. + +2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-server.c: + rtsp-server: Do not crash on nmapping of server + * generate error when gst_rtsp_connection_accept fails + * do not stop accepting incoming connections because + accepting a client fails + https://bugzilla.gnome.org/show_bug.cgi?id=701072 + +2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6 + https://bugzilla.gnome.org/show_bug.cgi?id=700953 + +2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com> + + * gst/rtsp-server/rtsp-sdp.c: + rtsp-sdp: Parse framerate caps field and set SDP attribute + The SDP attribute and its format is described in RFC4566. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747 + +2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com> + + * gst/rtsp-server/rtsp-sdp.c: + rtsp-sdp: Parse width/height from caps and set SDP attribute + The SDP attribute and its format is described in RFC6064. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747 + +2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-sdp.c: + * tests/check/gst/client.c: + rtsp-sdp: add bandwidth line + https://bugzilla.gnome.org/show_bug.cgi?id=699220 + +2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * common: + Automatic update of common submodule + From 5edcd85 to 098c0d7 + +2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com> + + * tests/check/gst/media.c: + tests: add dynamic payloader prepare/unprepare check + +2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: release lock when removing fakesink + +2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + stream: set elements to NULL before removing + When removing a stream, set the elements to NULL first. This avoids + element-is-not-in-NULL-state errors when we dispose the elements. + +2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net> + + * common: + Automatic update of common submodule + From 3cb3d3c to 5edcd85 + +2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: listen to pad-removed signals + Listen to the pad-removed signal and remove the stream associated with the + removed pad. + Add signal to be notified of the removed pad. + Remove the fakesink in unprepare() + Fix signatures of the signal methods + +2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-sdp.c: + tests: add example of reusable pipelines + +2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add method to get the srcpad + +2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com> + + * tests/check/gst/media.c: + check: add media prepare/unprepare test + See https://bugzilla.gnome.org/show_bug.cgi?id=698376 + +2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-media.c: + media: disconnect from signal handlers in unprepare() + We connected to the pad-added and no-more-pads signals in prepare() so + we need to disconnect from them in unprepare(). + See https://bugzilla.gnome.org/show_bug.cgi?id=698376 + +2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-media.c: + media: don't free streams array + Don't free the streams array in the unprepare() method, they were not + added in prepare(). + See https://bugzilla.gnome.org/show_bug.cgi?id=698376 + +2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-media.c: + media: don't unref the pipeline in unprepare + Unprepare() should undo what prepare() does. Because the pipeline is + not created in prepare(), we should not unref it in unprepare() + +2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: clear session and caps for reuse + Set the session and caps to NULL after unref otherwise we might unref + them again later. + See https://bugzilla.gnome.org/show_bug.cgi?id=698376 + +2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-client.c: + client: send out teardown signal before tearing down + The advantage is that in the signal handler you get direct access to + information about what streams are about to get torn down (in the + GstRTSPClientState). + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686 + +2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: expose connection + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546 + +2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net> + + * common: + Automatic update of common submodule + From aed87ae to 3cb3d3c + +2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-media.h: + media: add method to get the base_time of the pipeline + Together with a shared clock, this base-time could eventually be sent to + the client so that it can reconstruct the exact running-time of the clock + on the server. + +2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-sdp.c: + media: add GstNetTimeProvider support + Add a property to let the media provide a GstNetTimeProvider for its clock. + Make methods to get the clock and nettimeprovider + Add a x-gst-clock property to the SDP with the IP and port number of the nettime + provider and also the current time of the clock. This should make it possible + for (GStreamer) clients to slave their clock to the server clock. + +2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net> + + * common: + Automatic update of common submodule + From 04c7a1e to aed87ae + +2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: wait for buffering to complete + Wait for buffering to complete before changing the state to the target state. + +2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: small cleanup + +2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com> + + * tests/check/gst/rtspserver.c: + tests: remove extra unref in test_setup_non_existing_stream + The unref is not needed anymore, teardown runs without it. + https://bugzilla.gnome.org/show_bug.cgi?id=696542 + +2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com> + + * tests/check/gst/rtspserver.c: + tests: GSocketService cleanup in test_bind_already_in_use + Use g_socket_service_stop so the rtspserver test stops listening for + incoming connections in test_bind_already_in_use. + https://bugzilla.gnome.org/show_bug.cgi?id=696541 + +2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-media-factory.c: + rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here + Instead use a GWeakRef which is safe to use + This is a known GLib bug, see: + https://bugzilla.gnome.org/show_bug.cgi?id=667145 + +2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-sdp.c: + * tests/check/gst/media.c: + * tests/check/gst/rtspserver.c: + rtsp-media/client: Reply to PLAY request with same type of Range + Remember the type of Range from the PLAY request and use the same type for + the reply. + +2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * tests/check/gst/client.c: + rtsp-client: expose uri + +2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/gst/mediafactory.c: + tests: Hold ref while creating second media + To test if the media aren't shared, make sure we keep the first one while creating a second + otherwise the same memory address may be reused. + +2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net> + + * configure.ac: + configure: remove out-of-date comment + +2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net> + + * .gitignore: + .gitignore: ignore more build files + +2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net> + + * tests/check/Makefile.am: + tests: use right _LIBS variable for gst-plugins-base libs + +2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/Makefile.am: + check: add librtp to libs + +2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/gst/rtspserver.c: + tests: Add test to check selecting a port the server will send from + +2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/gst/rtspserver.c: + tests: Make sure packets are actually received + +2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: Select unicast address from pool if appropriate + +2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-stream.c: + stream: Properties are always there in Gst 1.0 + +2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/gst/addresspool.c: + tests: Add tests for unicast addresses in pool + +2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-address-pool.c: + * tests/check/gst/addresspool.c: + address-pool: Verify that multicast addresses are used for multicast and vice-versa + +2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-address-pool.h: + * gst/rtsp-server/rtsp-stream.c: + * tests/check/gst/addresspool.c: + address-pool: Add unicast addresses + +2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com> + + * configure.ac: + * gst/rtsp-server/rtsp-server.c: + * tests/check/gst/rtspserver.c: + rtsp-server: Limit the number of threads per server instance + If we exceed the maximum, just round robin the clients over the existing + threads. + +2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-server.c: + rtsp-server: No need to store the GMainContext in the client context + +2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/gst/rtspserver.c: + tests: Add test for client disconnection + +2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/gst/rtspserver.c: + tests: Test client and session timeouts with multiple threads + +2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + Document locking and its order + +2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/gst/rtspserver.c: + tests: Test that slow DESCRIBE don't block other clients + +2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/gst/client.c: + tests: Add tests for client-requested multicast address + +2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com> + + * docs/libs/gst-rtsp-server-sections.txt: + docs: Put the various functions in the right sections + +2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com> + + * docs/libs/gst-rtsp-server-docs.sgml: + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-address-pool.h: + docs: Generate docs for GstRTSPAddressPool + +2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + client: Check client provided addresses against the address pool + +2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-address-pool.h: + * tests/check/gst/addresspool.c: + address-pool: Add API to request a specific address from the pool + Also add relevant unit tests. + +2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/gst/mediafactory.c: + tests: Check the passing around of a RTSPAddressPool + Make sure the RTSPAddressPool is propagated from the MediaFactory all the + way down to the stream. + +2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com> + + * tests/check/gst/addresspool.c: + tests: Add more tests for the address pool + +2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com> + + * gst/rtsp-server/rtsp-address-pool.c: + address-pool: Fix off by one error + When splitting a port range, the port after a skip is not part of range. + +2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net> + + * common: + Automatic update of common submodule + From 2de221c to 04c7a1e + +2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com> + + * configure.ac: + configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS + AM_CONFIG_HEADER was removed in automake 1.13 + https://bugzilla.gnome.org/show_bug.cgi?id=693368 + +2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net> + + * common: + Automatic update of common submodule + From a942293 to 2de221c + +2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: make sure the watch exists while sending data + Protect the send_func with a lock. This allows us to wait for sending + to complete before changing the send_func and user_data. We add an + extra ref to the watch to make sure that it remains valid during + sending. + When closing the connection, set the send_func to NULL + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433 + +2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk> + + * tests/check/Makefile.am: + tests: use GST_*_1_0 environment variables everywhere + The _1_0 suffixed environment variables override the + non-suffixed ones, so if we're in an environment that + sets the _1_0 suffixed ones, such as jhbuild, we need + to set those to make sure ours actually always get + used. + +2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk> + + * common: + Automatic update of common submodule + From acb04d9 to a942293 + +2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: set the client backlog + Set the client backlog to a reasonable default + +2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Make the element a constructor parameter + https://bugzilla.gnome.org/show_bug.cgi?id=689594 + +2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com> + + * docs/libs/Makefile.am: + docs: Link with gcov library when gcov is enabled + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583 + +2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: match prepare with unprepare + Really unprepare when there were an equal amount of prepare calls. + +2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: media has to be unprepared in finalize + Because unprepare takes away the last ref on the media. + +2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it" + This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05. + We can't use the refcount to trigger unprepare because it is the unprepare call + that removes the last refcount after all messages are consumed. What we should + probably do is make a prepared refcount and only unprepare when the refcount + reaches 0. + +2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: let the source unref the last media ref + the last ref to the media is held by the source so we don't need to add more ref + and unrefs, we simply destroy the media when the source is gone. + +2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: improve debug + +2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: check state + Make sure we are in the right state when collecting the position and duration. + Only make ourselves PREPARED when we were previously PREPARING. + +2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: use g_object_ref/unref for GObjects + +2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com> + + * gst/rtsp-server/rtsp-client.c: + client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it + Calling gst_rtsp_media_unprepare breaks shared medias. Just unref + GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media + isn't being used anymore. + +2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com> + + * gst/rtsp-server/rtsp-media.c: + Fix compiler warning + +2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com> + + * gst/rtsp-server/rtsp-media-factory-uri.c: + Add missing g_type_class_add_private in GstRTSPMediaFactoryURI + +2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-session-media.h: + small cleanup + +2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * tests/check/gst/media.c: + media: avoid element leak + +2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: require an element in media constructor + +2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + Revert "client: TEARDOWN brings that state to Init again" + This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e. + The object is already disposed, there is no point in setting the state. + +2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: TEARDOWN brings that state to Init again + +2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-sections.txt: + * examples/test-auth.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory-uri.h: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-mount-points.h: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-media.h: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session-pool.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/media.c: + rtsp: make object details private + Make all object details private + Add methods to access private bits + +2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/Makefile.am: + * tests/check/gst/media.c: + tests: add media tests + +2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: check if prepared for some methods + Check that the media object is prepared before doing seek and getting the + current position etc. + Add some g_return checks. + +2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/Makefile.am: + * tests/check/gst/mediafactory.c: + tests: add mediafactory test + +2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + stream: improve debug + +2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: unref pipeline in finalize to avoid leaking it + +2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media.c: + rtsp: use gst_object_unref on GstObjects + +2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: require an url + +2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-uri.c: + examples: fix include + +2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.h: + server: remove unused include + +2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/Makefile.am: + * tests/check/gst/mountpoints.c: + tests: add test for mountpoints + +2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: fix factory leak + Keep the factory in the state object only for authorization checks and make + sure we unref it on failure. Also don't keep invalid objects in the state + object. + +2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-mount-points.c: + mounts: add g_return_if guards + +2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/gst/client.c: + tests: add more tests + +2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: improve debug + +2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: improve debug and fix leaks + Cleanup the uri and session when there is a bad request. + +2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * common: + update common + +2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/gst/client.c: + test: add test for session in options request + +2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: use 454 when session can't be found + We should use 454 when a session can't be found because there was no session + pool configured in the server. This is not a server configuration problem + because the server on which the request is done might not be the same one that + will keep the sessions for us and so it does not need to support sessions. + +2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: only free connection when there is one + It's possible that the client doesn't have a connection when we try to free it. + +2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/Makefile.am: + * tests/check/gst/client.c: + tests: add unit test for the client object + +2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: small cleanup + +2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.h: + client: remove unused include + +2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: fix compilation + +2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: call destroy without the lock + +2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: make the client usable without a socket + Make a method to let the client handle a message and a callback when the client + wants us to send a response message back. This makes it possible to also use the + client object without the sockets, which should make it easier to test. + +2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: small cleanup + +2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-server.c: + client: remove reference to server + We don't need to keep a ref to the server + +2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: add locking + Also add some g_return_if() + +2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: log more errors + +2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: fix compilation + +2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: add generic close-after-send support + Add a property to send_response() to close the connection after the response has + been sent to the client. + +2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/README: + * docs/libs/gst-rtsp-server-docs.sgml: + * docs/libs/gst-rtsp-server-sections.txt: + * docs/libs/gst-rtsp-server.types: + * examples/test-auth.c: + * examples/test-launch.c: + * examples/test-mp4.c: + * examples/test-multicast.c: + * examples/test-multicast2.c: + * examples/test-ogg.c: + * examples/test-readme.c: + * examples/test-sdp.c: + * examples/test-uri.c: + * examples/test-video.c: + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media-mapping.c: + * gst/rtsp-server/rtsp-media-mapping.h: + * gst/rtsp-server/rtsp-mount-points.c: + * gst/rtsp-server/rtsp-mount-points.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session-pool.h: + * tests/check/gst/rtspserver.c: + MediaMapping -> MountPoints + Describes better what the object manages. + +2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * configure.ac: + configure: bump required version of -base + +2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: fix seeking + +2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: support more Range formats + Use the new -base methods to convert the Range string into a seek start and stop + value. + +2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-launch.c: + examples: fix whitespace + +2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-auth.c: + test-auth: add example of how to remove sessions + Add an example of the session filter api. + +2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-uri.c: + test-uri: remove mapping example + +2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-uri.c: + test-uri: fix callback signature + +2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-factory.c: + factory: keep ref to factory while media active + While the media from a factory is alive, keep a ref to the factory. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555 + +2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-factory-uri.c: + factory-uri: add some debug + +2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + stream: set udp sources to PLAYING + Set the UDP sources to PLAYING and locked state before we add it to the pipeline + so that it doesn't cause our pipeline to produce ASYNC-DONE. + +2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-factory-uri.c: + factory-uri: take ref to factory + Take a ref to the factory that we place in our list. + +2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/Makefile.am: + * tests/test-reuse.c: + test: add test for server reuse + See https://bugzilla.gnome.org/show_bug.cgi?id=688395 + +2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-server.c: + server: start and stop multiple times + Stop listening on the RTSP port when the GSource is removed, so clients + can't connect and the server can be started again. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395 + +2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + server: fix small leak + +2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: unref source in finish_unprepare + The source is created in prepare, unref it in finish_unprepare. + See https://bugzilla.gnome.org/show_bug.cgi?id=688707 + +2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + rtsp-media: remove bus watch before finalizing + * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare. + * An extra media ref is added for the bus watch. This extra ref is unreffed by + the GDestroyNotify function. + * gst_rtsp_media_unprepare destroys the source so the bus watch is removed. + * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls + gst_rtsp_media_unprepare before unreffing the media. + This way, the bus watch will be removed before the media is finalized. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707 + +2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: wait until the TEARDOWN response is sent to close the connection + Responses can be sent async so we need to wait until the TEARDOWN response has + been written before we close the connection to the client. This avoids the risk + of writing/polling closed sockets. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535 + +2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: plug socket leak + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703 + +2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net> + + * common: + Automatic update of common submodule + From 6bb6951 to a72faea + +2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net> + + * gst/rtsp-server/rtsp-media-factory-uri.c: + rtsp-server: don't use deprecated API + +2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: fix unused-but-set-variable compiler warning + rtsp-client.c:1260:21: error: variable 'protocols' set but not used + +2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * TODO: + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-client.c: + rtsp: cleanups + +2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/Makefile.am: + * examples/test-multicast2.c: + examples: add another multicast example + Add an example for how to configure separate multicast ranges for each media + stream. + +2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-multicast.c: + test: set shared + +2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-media.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + stream: use the address managed by the stream + Use the address managed by the stream for multicast. This allows us to have 1 + multicast address for each stream. + Because the address is now managed by the stream we don't have to pass it around + anymore. + Set the address pool on the streams. + +2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + rtsp: improve debug + +2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add signal for new streams + This allows applications to listen for new streams and configure properties on + them, like the address pool. + +2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: configure address pool in new streams + +2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add methods to deal with address pool + Add methods to get and set the address pool for the stream + Add method to allocate and get the multicast addresses for this stream. + +2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: remove MTU property + It is a stream property + +2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: set blocksize only on stream + Set the blocksize only on the current stream. + +2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + stream: share src and sink sockets + the allocated socket is in the used-socket property, not socket. + +2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-address-pool.h: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-media.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + * tests/check/gst/addresspool.c: + rtsp: make address-pool return an address object + Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to + store more info in the structure and allows us to more easily return the address + to the right pool when no longer needed. + Pass the address to the StreamTransport so that we can return it to the pool + when the stream transport is freed or changed. + +2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/Makefile.am: + * examples/test-multicast.c: + examples: add multicast example + Show how to set up the multicast address pool so that media can be + server with multicast. + +2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + rtsp: use AddressPool + Remove the multicast_group property. + Use the configured addresspool to allocate multicast addresses. + +2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-address-pool.h: + address-pool: add clear method + +2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-address-pool.c: + address-pool: small cleanups + +2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * tests/check/Makefile.am: + * tests/check/gst/addresspool.c: + tests: add addresspool unit test + +2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-address-pool.h: + address-pool: add object to manage multicast addresses + Make an object that can manage a rage of multicast addresses and ports. + +2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + server: set default max-threads property + +2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: wait for concurrent _prepare + If a prepare is busy, wait for the result. + +2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: add lock around message handler + We don't want to dispatch messages while we are still processing the result of + the state change. + +2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add lock to protect state changes + +2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add locking + +2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + * gst/rtsp-server/rtsp-stream.c: + stream-transport: add keep-alive method + +2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + * gst/rtsp-server/rtsp-stream.c: + stream-transport: add method to handle RTP/RTCP + Call new methods instead of poking into the structures directly. + +2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-media.h: + session-media: add locking + +2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + session: add locking + +2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + server: free old socket + +2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-mapping.c: + * gst/rtsp-server/rtsp-media-mapping.h: + mapping: add locking + +2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: add locking + +2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + auth: add locking + +2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: add max-thread property + +2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: use a threadpool for the mainloops + +2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: rename method + gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we + don't really create the client from the socket, we use the socket for the + client. + +2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-server.c: + server: rework maincontext handling in clients + Make a separate method to attach a client to a MainContext. + Let the server decide in what GMainContext the client will operate and give this + context to the client in attach. Then the server can later decide to use a + separate thread for each client or just use the mainthread. + +2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + session: move session header code in session object + +2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net> + + * COPYING: + * COPYING.LIB: + * examples/test-auth.c: + * examples/test-launch.c: + * examples/test-mp4.c: + * examples/test-ogg.c: + * examples/test-readme.c: + * examples/test-sdp.c: + * examples/test-uri.c: + * examples/test-video.c: + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory-uri.h: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media-mapping.c: + * gst/rtsp-server/rtsp-media-mapping.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-params.c: + * gst/rtsp-server/rtsp-params.h: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-sdp.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-media.h: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session-pool.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + * tests/check/gst/rtspserver.c: + * tests/test-cleanup.c: + Fix FSF address + +2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session.c: + rtsp-server: added annotations to indicate type of ownership transfer of return values + https://bugzilla.gnome.org/show_bug.cgi?id=680777 + +2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net> + + * configure.ac: + No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now + +2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net> + + * Makefile.am: + * bindings/Makefile.am: + * bindings/vala/Makefile.am: + * bindings/vala/gst-rtsp-server-0.10.deps: + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.deps: + * bindings/vala/packages/gst-rtsp-server-0.10.files: + * bindings/vala/packages/gst-rtsp-server-0.10.gi: + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + * bindings/vala/packages/gst-rtsp-server-0.10.namespace: + * configure.ac: + bindings: remove vala bindings + They'll be reunited with the other GStreamer bindings + https://bugzilla.gnome.org/show_bug.cgi?id=680777 + +2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-media.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + rtsp: only create transport when needed + Only create the StreamTransport when configured. + +2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: small cleanup + +2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + rtsp: refactor configuration of transport + Move the configuration of the transport to a place where it makes + more sense. + +2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: refactor transport parsing + +2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: refuse to change the MTU on shared media + If we change the MTU of chared media, it changes for all clients. + We don't want to set the MTU to something large for clients that + stream over UDP. + +2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * examples/test-mp4.c: + * gst/rtsp-server/rtsp-media.c: + small fixes to docs and debug + +2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-stream.c: + stream: transports must already have been removed + +2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: improve join and leave of the pipeline + simplify code + Do the cleanup properly + Add some docs + +2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: move unprepare below default implementation + Makes it easier to find the default implementation + +2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: signal unprepared when we actually finish + +2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-media.c: + media: no need to unlock, unprepare does that when needed + +2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/libs/gst-rtsp-server-sections.txt: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media-mapping.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-params.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-session-pool.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + * gst/rtsp-server/rtsp-stream-transport.h: + * gst/rtsp-server/rtsp-stream.h: + docs: update docs + +2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-mapping.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + rtsp: fix MTU setting + Fix setting of the MTU. There is no need for a vmethod. + +2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * docs/README: + docs: update docs + +2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> + + * configure.ac: + configure: bump version number after refactoring + +2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-media.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream-transport.h: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + rtsp: massive refactoring + Make GObjects from the remaining simple structures. + Remove GstRTSPSessionStream, it's not needed. + Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter + Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how + a GstRTSPStream should be transported to a client. + Rename GstRTSPMediaFactory::get_element -> create_element because that + more accurately describes what it does. + Make nice methods instead of poking in the structures. + Move some methods inside the relevant object source code. + Use GPtrArray to store objects instead of plain arrays, it is more + natural and allows us to more easily clean up. + Move the allocation of udp ports to the Stream object. The Stream object + contains the elements needed to stream the media to a client. + Improve the prepare and unprepare methods. Unprepare should now undo + everything prepare did. Improve also async unprepare when doing EOS on + shutdown. Make sure we always unprepare correctly. + +2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Unref server address clients connected to + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725 + +2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-server.c: + rtsp-server: don't ref server socket if it is NULL + Fixes test_bind_already_in_use unit test again after commit 6a497440. + https://bugzilla.gnome.org/show_bug.cgi?id=686644 + +2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com> + + * tests/check/Makefile.am: + tests: Add libgio link dependency + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647 + +2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org> + + * gst/rtsp-server/rtsp-media-mapping.c: + * gst/rtsp-server/rtsp-media-mapping.h: + rtsp-media-mapping: rename find_media vfunc to find_factory + The virtual method and class method should have the same name + so it is correctly represented in GIR file + https://bugzilla.gnome.org/show_bug.cgi?id=680777 + +2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-mapping.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + rtsp-server: fixed comments and GIR annotations + https://bugzilla.gnome.org/show_bug.cgi?id=680777 + +2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com> + + * gst/rtsp-server/rtsp-media-mapping.c: + media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory + +2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com> + + * gst/rtsp-server/rtsp-server.c: + rtsp-server: allow binding on port 0 (binds on a random port) + +2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com> + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + rtsp-server: add bound-port property + bound-port can be used to retrieve the port number when the server is bound on + port 0, which binds on a random port. + +2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com> + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + rtsp-media-factory: make ::get_element overridable by GI bindings + The way to annotate vfuncs with GI seems to be to create an invoker (GI term) + for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element() + as the invoker for ::get_element(), making it overridable by GI generated + bindings. + +2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com> + + * gst/rtsp-server/rtsp-media-factory-uri.c: + rtsp-media-factory-uri: don't autoplug parsers in a loop + Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging + h264parse forever. + +2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com> + + * gst/rtsp-server/Makefile.am: + Explicitly link against gio. Fix link error on mac. + +2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com> + + * gst/rtsp-server/rtsp-session.c: + session: add ttl to the transport header in SETUP + See https://bugzilla.gnome.org/show_bug.cgi?id=685561 + +2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media.c: + client: Use client transport settings for multicast if allowed. + This patch makes it possible for the client to send transport settings for + multicast (destination && ttl). Client settings must be explicitly allowed or + the server will use its own settings. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561 + +2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net> + + * common: + Automatic update of common submodule + From 6c0b52c to 6bb6951 + +2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: do not destroy the rtsp watch + Don't destroy the client watch while dispatching. The rtsp watch is + automatically destroyed after the rtsp watch function closed() has + been called. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220 + +2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net> + + * common: + Automatic update of common submodule + From 4f962f7 to 6c0b52c + +2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-media.c: + media: fix check for seekability + +2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: use more GIO + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593 + +2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-server.c: + server: remove obsolete includes + +2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com> + + rtsp-media: also initialize transports in on_ssrc_active (bug #683304) + * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not + be available in "on_new_ssrc". The transports are added in + gst_rtsp_media_set_state when going to PLAYING state. However, + "on_new_ssrc" might be called before this happens. + https://bugzilla.gnome.org/show_bug.cgi?id=683304 + +2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + rtsp-client: add signals for rtsp requests (fixes #683287) + +2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + add new-session signal to rtsp-client (fixes #683058) + +2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net> + + * common: + Automatic update of common submodule + From 668acee to 4f962f7 + +2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-server.c: + * tests/check/gst/rtspserver.c: + rtsp-server: fixed segfault in gst_rtsp_server_create_socket + Do not assume that *error is set in g_socket_address_enumerator_next. + Added test_bind_already_in_use unit-test. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914 + +2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net> + + * common: + Automatic update of common submodule + From 94ccf4c to 668acee + +2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + rtsp-client: make create_sdp virtual method + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173 + +2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * common: + Automatic update of common submodule + From 98e386f to 94ccf4c + +2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: fix docs + +2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + rtsp-server: use an existing socket to establish HTTP tunnel + Make it possible to transfer a socket from an HTTP server to be used as + an RTSP over HTTP tunnel. + +2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + rtsp: Handle the blocksize parameter + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325 + +2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com> + + * tests/check/Makefile.am: + * tests/check/gst/rtspserver.c: + Have unit test get header from source dir, not installed dir + This makes compilation of unit tests work in a build directory other + than the source directory. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789 + +2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: update for gst_element_make_from_uri() changes + +2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com> + + * configure.ac: + * tests/Makefile.am: + * tests/check/Makefile.am: + * tests/check/gst/rtspserver.c: + rtsp: add unit test + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076 + +2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: don't collect media stats when going to NULL + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015 + +2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + client: don't leak transports + +2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: free transport on no_stream in SETUP handler + +2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: changed session media iteration + In client_unlink_session: now don't iterate in session->medias + list where items are removed by gst_rtsp_session_release_media. + Instead, repeatedly remove the first item. + +2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: don't use g_object_unref on GstRTSPSessionMedia + GstRTSPSessionMedia is not a GObject type. When the + GstRTSPSession is freed, it will free the media. + +2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com> + + * gst/rtsp-server/rtsp-media-factory.c: + factory: plug pad leak in collect_streams + In gst_rtsp_media_factory_collect_streams: unref the srcpad that + was retrieved using gst_element_get_static_pad. gst_ghost_pad_new + will take one reference, and the other reference will otherwise + give a memory leak. + +2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com> + + * configure.ac: + configure: suppress some warnings when debug is disabled + Warnings about unused variables should be suppressed if core has the + debug system disabled. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824 + +2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> + + * docs/libs/Makefile.am: + docs: fix build in uninstalled setup + Include gst-plugins-base libs properly. + +2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com> + + * docs/libs/gst-rtsp-server.types: + docs: include headers defining rtsp-server object types + Fixes compiler warnings during docs build. + https://bugzilla.gnome.org/show_bug.cgi?id=676824 + +2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com> + + * configure.ac: + configure: Add warning flags for compiler when configuring + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824 + +2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk> + + * common: + Automatic update of common submodule + From 03a0e57 to 98e386f + +2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk> + + * common: + Automatic update of common submodule + From 1fab359 to 03a0e57 + +2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com> + + * gst/rtsp-server/rtsp-client.c: + client: fix GSocketAddress leak in gst_rtsp_client_accept + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463 + +2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk> + + * common: + Automatic update of common submodule + From f1b5a96 to 1fab359 + +2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * common: + Automatic update of common submodule + From 92b7266 to f1b5a96 + +2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * common: + Automatic update of common submodule + From ec1c4a8 to 92b7266 + +2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * common: + Automatic update of common submodule + From 3429ba6 to ec1c4a8 + +2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-server.c: + rtsp: fix compiler warnings + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500 + +2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * common: + Automatic update of common submodule + From dc70203 to 3429ba6 + +2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk> + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session-pool.h: + rtsp-server: port to new thread API + +2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * common: + Automatic update of common submodule + From 6db25be to dc70203 + +2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + rtsp-server: Fix compilation and compiler warnings + +2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * autogen.sh: + * configure.ac: + * gst/rtsp-server/Makefile.am: + configure: Modernize autotools setup a bit + Also we now only create tar.bz2 and tar.xz tarballs. + +2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * common: + Automatic update of common submodule + From 464fe15 to 6db25be + +2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk> + + * common: + Automatic update of common submodule + From 7fda524 to 464fe15 + @@ -1 +1,19 @@ -This is GstRTSP +This is the Gstreamer RTSP Server Library 1.2.3 + +Changes since 0.10.8: + + * Ported to GStreamer 1.0 API + * Multithreaded for better responsiveness and scaleability + * Support for authentication and access restrictions + * Support for TLS + * API refactored a bit for clarity (e.g. MediaMapping -> MountPoints) + * Improved seeking support + * Better live pipeline support + * Better IPv6 and multicast support + * Improved portability (ported to gio) + * Support for automatic synchronous playback of multiple clients via GstNetTimeProvider + * Support for more range formats + * New example how to feed data to the server via appsrc + * New example how to prioritise some clients over others using cgroups + * Many more unit tests + * Hundreds of bug-fixes and other improvements @@ -0,0 +1,85 @@ + +Release notes for GStreamer RTSP Server Library 1.2.3 + + +The GStreamer team is proud to announce the first release +of the GStreamer RTSP Server Library for the GStreamer 1.0 +API series. + + +Features of this release + + * Ported to GStreamer 1.0 API + * Multithreaded for better responsiveness and scaleability + * Support for authentication and access restrictions + * Support for TLS + * API refactored a bit for clarity (e.g. MediaMapping -> MountPoints) + * Improved seeking support + * Better live pipeline support + * Better IPv6 and multicast support + * Improved portability (ported to gio) + * Support for automatic synchronous playback of multiple clients via GstNetTimeProvider + * Support for more range formats + * New example how to feed data to the server via appsrc + * New example how to prioritise some clients over others using cgroups + * Many more unit tests + * Hundreds of bug-fixes and other improvements + +==== Download ==== + +You can find source releases of gst-rtsp-server in the download +directory: http://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/ + +==== Support and Bugs ==== + +We use GNOME's bugzilla for bug reports and feature requests: +http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer + +Please submit patches via bugzilla as well. + +For help and support, please subscribe to and send questions to the +gstreamer-devel mailing list (see below for details). + +There is also a #gstreamer IRC channel on the Freenode IRC network. + +==== Developers ==== + +GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned +from there (see link above). + +Interested developers of the core library, plugins, and applications should +subscribe to the gstreamer-devel list. + + +Contributors to this release + + * Aleix Conchillo Flaqué + * Alessandro Decina + * Alexander Schrab + * Andrey Utkin + * Branko Subasic + * David Svensson Fors + * Edward Hervey + * George McCollister + * Göran Jönsson + * Jonas Holmberg + * Lubosz Sarnecki + * Ognyan Tonchev + * Olivier Crête + * Patricia Muscalu + * Patrick Radizi + * Sebastian Dröge + * Sebastian Pölsterl + * Sebastian Rasmussen + * Stefan Sauer + * Tim-Philipp Müller + * Wim Taymans + * Youness Alaoui + diff --git a/configure.ac b/configure.ac index 9abec60..08487e2 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ AC_PREREQ(2.62) 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], [1.1.90], +AC_INIT([GStreamer RTSP Server Library], [1.2.3], [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer], [gst-rtsp-server]) AG_GST_INIT @@ -53,7 +53,7 @@ dnl 1.2.5 => 205 dnl 1.10.9 (who knows) => 1009 dnl dnl sets GST_LT_LDFLAGS -AS_LIBTOOL(GST, 200, 0, 200) +AS_LIBTOOL(GST, 203, 0, 203) dnl *** required versions of GStreamer stuff *** GST_REQ=1.2.3 diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap index c2e925e..9f4b370 100644 --- a/gst-rtsp-server.doap +++ b/gst-rtsp-server.doap @@ -32,6 +32,16 @@ RTSP server library based on GStreamer <release> <Version> + <revision>1.2.3</revision> + <branch>1.2</branch> + <name></name> + <created>2014-02-25</created> + <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.2.3.tar.xz" /> + </Version> + </release> + + <release> + <Version> <revision>1.1.90</revision> <branch>1.1</branch> <name></name> |